{"owner":"wwmm","repo":"easyeffects","hasSkills":true,"totalSkillsCount":33,"totalTokensCount":17653,"categories":["anthropic-skill"],"hasMcp":false,"mcpConfig":null,"found":["src/contents/docs/plugins/autogain.md","src/contents/docs/plugins/autotune.md","src/contents/docs/plugins/bassenhancer.md","src/contents/docs/plugins/bassloudness.md","src/contents/docs/plugins/compressor.md","src/contents/docs/plugins/convolver.md","src/contents/docs/plugins/crossfeed.md","src/contents/docs/plugins/crosstalkcanceller.md","src/contents/docs/plugins/crusher.md","src/contents/docs/plugins/crystalizer.md","src/contents/docs/plugins/deepfilternet.md","src/contents/docs/plugins/deesser.md","src/contents/docs/plugins/delay.md","src/contents/docs/plugins/echocanceller.md","src/contents/docs/plugins/equalizer.md","src/contents/docs/plugins/exciter.md","src/contents/docs/plugins/expander.md","src/contents/docs/plugins/filter.md","src/contents/docs/plugins/gate.md","src/contents/docs/plugins/harmonizer.md","src/contents/docs/plugins/limiter.md","src/contents/docs/plugins/loudness.md","src/contents/docs/plugins/maximizer.md","src/contents/docs/plugins/midside_equalizer.md","src/contents/docs/plugins/multibandcompressor.md","src/contents/docs/plugins/multibandgate.md","src/contents/docs/plugins/pitch.md","src/contents/docs/plugins/reverb.md","src/contents/docs/plugins/rnnoise.md","src/contents/docs/plugins/spectrum.md","src/contents/docs/plugins/speex.md","src/contents/docs/plugins/stereotools.md","src/contents/docs/plugins/voicesuppressor.md"],"skills":{"src/contents/docs/plugins/autogain.md":"# Auto Gain\n\nEasy Effects Autogain is based on the library libebur128 which implements the EBU R 128 standard for loudness normalization. It changes the audio volume to a perceived loudness target that can be customized by the user.\n\n**Target** \nLoudness level.\n\n**Silence** \nSilence threshold. While the momentary loudness is below this value the autogain won't make any changes to the current gain being applied to the signal level.\n\n**Maximum History** \nRange of time taken into account for the calculation of loudness level and output gain.\n\n**Reference** \nThe parameter used as reference to evaluate the output gain.\n\n- **Momentary** - Measures the loudness of the past 400 milliseconds.\n- **Short-Term** - Measures the loudness of the past 3 seconds.\n- **Integrated** - Ideally it indicates how loud the content is on average. It measures the loudness on a long range of time, depending on the value set as Maximum History.\n- **Geometric Mean** - Uses the geometric mean of all the above mentioned parameters, or two of them at user choosing.\n\n**Reset History** \nResets the Autogain history related to chosen Reference.\n\n**Monitor Parameters** \nAutogain values shown as stats.\n\n- **Relative** - Used to detect silence. Whenever the Momentary term is below a predetermined threshold, modifications to the output gain will be disabled.\n- **Loudness** - The difference between its value and the target loudness determines the output gain.\n- **Range** - Indicates how large is the dynamic range of the content played.\n- **Output Gain** - The input signal is adjusted by this correction gain to bring its loudness to the target value.\n\n## References\n\n- [Wikipedia EBU R 128](https://en.wikipedia.org/wiki/EBU_R_128)\n- [EBU - Loudness Normalisation and Permitted Maximum Level of Audio Signals](https://tech.ebu.ch/publications/r128/)\n","src/contents/docs/plugins/autotune.md":"# Autotune\n\nAutotune is a pitch correction effect that automatically adjusts the pitch of an audio signal to the nearest target note. It can be used to subtly correct vocal intonation or to create the characteristic \"auto-tune effect\" heard in modern music. Easy Effects uses the fat1 plugin from x42 LV2 Plugins.\n\n## Tuning Options\n\n**Mode** \nControls how pitch correction is applied.\n\n- **Auto** - Automatically detects the pitch and corrects it to the nearest enabled note in the scale.\n- **Manual** - Targets a fixed pitch set by the Offset parameter.\n\n**Tuning** \nThe reference frequency for the A4 note, in Hz. The standard tuning is 440 Hz. Adjust this if you are working with non-standard tuning (range: 400–480 Hz).\n\n**Correction** \nControls the strength of pitch correction. At 1.0, the pitch is fully corrected to the target note. Lower values allow more of the natural pitch variation through, producing a subtler effect.\n\n**Bias** \nControls the preference toward the current note versus the nearest target note. At 0.5 (center), no bias is applied. Higher values make the correction stick to the current note longer before switching. Lower values make it switch to the nearest target more readily.\n\n**Filter** \nControls the smoothness of the pitch detection. Lower values result in faster, more responsive detection but may introduce artifacts. Higher values produce smoother, more stable detection but with slower response.\n\n**Offset** \nShifts the target pitch by the specified number of semitones (range: -2.0 to 2.0). In Manual mode, this sets the fixed target pitch offset.\n\n**Fast Correction** \nWhen enabled, applies pitch correction with minimal latency. This produces a more immediate, aggressive correction effect but may introduce more artifacts.\n\n## Scale Selection\n\nThe Scale section allows you to select which notes the pitch correction will target.\n\n**Root** \nThe root note of the scale (C, D, E, F, G, A, or B).\n\n**Accidental** \nApplies a sharp or flat modifier to the root note.\n\n**Scale** \nSelects either Major or Minor scale pattern. Choosing a scale automatically enables the correct notes for that key.\n\n**Note Toggles** \nIndividual toggles for each of the 12 chromatic notes (C through B). Selecting a scale sets these automatically, but you can also configure them manually for custom scales or chromatic correction (all notes enabled).\n\n## Pitch Error Meter\n\nDisplays the current pitch deviation from the target note. A value of 0 means the pitch is exactly on target. Negative values indicate the pitch is flat, positive values indicate it is sharp.\n\n## References\n\n- [x42 fat1 Auto Tune](https://x42-plugins.com/x42/x42-autotune)\n- [Wikipedia Auto-Tune](https://en.wikipedia.org/wiki/Auto-Tune)\n","src/contents/docs/plugins/bassenhancer.md":"# Bass Enhancer\n\nA Bass Enhancer is used to produce very low sound that is not present in the original signal. This is done by creating harmonic distortions of the signal which are restricted in range and added to the original signal. It raises the lower end of an audio signal without simply raising the lower frequencies like an [Equalizer](equalizer.md) would do to create a more \"fat\" or \"boomy\" sound.\n\n**Amount** \nAmount of harmonics added to the original signal.\n\n**Harmonics** \nAmount of newly created harmonics.\n\n**Scope** \nThe frequency above which harmonics are produced.\n\n**Floor (button)** \nConstrain the enhancement on the lower end.\n\n**Floor (value)** \nThe frequency below which no harmonics are produced.\n\n**Blend Harmonics** \nThe \"colour\" (or octave) of the harmonics.\n\n**Listen** \nMute the original signal and listen to the harmonics exclusively.\n\n## References\n\n- [Calf Bass Enhancer](https://calf-studio-gear.org/doc/Bass%20Enhancer.html)\n- [Wikipedia Missing Fundamental](https://en.wikipedia.org/wiki/Missing_fundamental)\n","src/contents/docs/plugins/bassloudness.md":"# Bass Loudness\n\nThe ear is less sensitive to low frequencies when listening at low volume. This plugin developed by MDA allows the bass level to be adjusted measuring the \"equal-loudness contour\".\n\n**Loudness** \nSource level relative to listening level (based on a 100 dB SPL maximum level).\n\n**Output** \nChange output level.\n\n**Link** \nAutomatically adjusts Output to maintain a consistent tonal balance at all levels.\n\n## References\n\n- [Wikipedia Equal-Loudness Contour](https://en.wikipedia.org/wiki/Equal-loudness_contour)\n- [MDA Loudness](http://mda.smartelectronix.com/vst/help/loudness.htm)\n","src/contents/docs/plugins/compressor.md":"# Compressor\n\nA Compressor is used to reduce the dynamic range or, in other words, the difference in level between the quietest and the loudest parts of an audio signal. It achieves this purpose altering the gain when the signal overtakes a predetermined Threshold. Easy Effects uses the Stereo Sidechain Compressor from Linux Studio Plugins.\n\n## Compressor Options\n\n**Attack Time** \nThe length of time it takes to apply roughly two-thirds of the targeted amount of compression ratio to the uncompressed signal.\n\n**Release Time** \nThe length of time it takes to restore roughly two-thirds of the reduced gain (in Downward mode) or increased gain (in Upward/Boosting mode) to the compressed signal.\n\n**Attack Threshold** \nThe target level around which the compression is applied (the range depends by the Knee).\n\n**Release Threshold** \nSets up the Threshold of the Release Time, calculated by summing the Release Threshold to the Attack Threshold.\n\nIf the Sidechain level is above that Threshold, the compressor uses the Release Time for the releasing stage. Otherwise the Attack Time is used in place of Release Time.\n\nFor example, with -10 dB Attack Threshold and -60 dB Release Threshold, if the Sidechain is above `-10 + (-60) = -70 dB`, the Release Time is used for the gain restoration. If the Sidechain is below -70 dB, the Attack Time is used for the gain restoration.\n\n**Ratio** \nThe amount of attenuation (in Downward mode) or amplification (in Upward/Boosting mode) that will be applied to the signal.\n\nFor example, when the Ratio is 2 in Downward mode and the Sidechain rises above the Threshold by 10 dB, the signal would be ideally reduced by 5 dB (10/2 dB). In practice this behavior mostly depends on how the Compressor is designed and configured.\n\n**Knee** \nThe range over which the Compressor switches from no compression to almost the full ratio compression (the Threshold usually sits at the center of this transition zone).\n\n**Makeup** \nThe gain to apply after the compression stage.\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal mixed in the output.\n\n**Mode** \n- **Downward** - It's aimed to decrease the gain of the signal above the Threshold.\n- **Upward** - It's aimed to increase the gain of the signal below the Threshold.\n- **Boosting** - It's aimed to increase the gain of the signal below the Threshold by a specific amount.\n\n**Boost Threshold** \nThe Threshold below which a constant amplification will be applied to the input signal in Upward Mode (it prevents from applying infinite amplification to very quiet signals).\n\n**Boost Amount** \nMaximum gain amplification to apply in Boosting Mode.\n\n## Sidechain\n\n**Listen** \nAllows to listen the processed Sidechain signal.\n\n**Input Type** \nDetermines which signal is the Sidechain or, in other words, the signal that controls the compression stage.\n\n- **Feed-forward** - The Sidechain is the Compressor input signal (taken after applying the plugin input gain). More aggressive compression.\n- **Feed-back** - The Sidechain is the Compressor output signal (taken before applying the Makeup and the plugin output gain). Vintage-style compression.\n- **External** - The Sidechain is an external source took by a specific input device (typically a microphone).\n\n**Input Device** \nSelect the device for the External Sidechain.\n\n**Mode** \nDetermines how the Sidechain is evaluated for the compression stage.\n\n- **Peak** - The Compressor reacts according to the peaks.\n- **RMS** - The Compressor reacts according to the average loudness measured by the root mean square.\n- **Low-Pass** - The Compressor reacts according to the signal processed by a Low-Pass filter.\n- **Uniform** - The Compressor reacts according to the loudness measured by the average of the absolute amplitude.\n\n**Source** \nDetermines which part of the Sidechain is taken into account for the compression stage.\n\n- **Middle** - The sum of left and right channels.\n- **Side** - The difference between left and right channels.\n- **Left** - Only left channel is used.\n- **Right** - Only right channel is used.\n- **Min** - The absolute minimum value is taken from stereo input.\n- **Max** - The absolute maximum value is taken from stereo input.\n\n**PreAmplification** \nGain applied to the Sidechain signal.\n\n**Reactivity** \nThe time that defines the number of samples used to process the Sidechain in RMS, Uniform and Low-Pass modes. Higher the value, more smooth the compression.\n\n**Lookahead** \nThe signal to compress is delayed by this amount of time, so that the compression will be applied earlier than it would be otherwise. The corresponding delay is reproduced on the output signal.\n\n## Sidechain Filters\n\n**High-Pass Filter Mode** \nSets the type of the High-Pass filter applied to Sidechain signal.\n\n**High-Pass Frequency** \nSets the cut-off frequency of the High-Pass filter.\n\n**Low-Pass Filter Mode** \nSets the type of the Low-Pass filter applied to Sidechain signal.\n\n**Low-Pass Frequency** \nSets the cut-off frequency of the Low-Pass filter.\n\n## References\n\n- [Wikipedia Dynamic Range Compression](https://en.wikipedia.org/wiki/Dynamic_range_compression)\n- [LSP Sidechain Compressor Stereo](https://lsp-plug.in/?page=manuals§ion=sc_compressor_stereo)\n- [Black Ghost Audio - The Ultimate Guide to Compression](https://www.blackghostaudio.com/blog/the-ultimate-guide-to-compression)\n- [Attack Magazine - Demolishing The Myths of Compression](https://www.attackmagazine.com/features/columns/gregory-scott-demolishing-the-myths-of-compression/)\n","src/contents/docs/plugins/convolver.md":"# Convolver\n\nThe Convolver creates a simulation of an audio environment using a pre-recorded audio sample of the impulse response of the space being modeled. This feature is based on the \"convolution\": a process through which the sonic characteristics of one signal are used to alter the character of another.\n\nEasy Effects Convolver offers the opportunity to apply multiple impulse responses by combining them in one file.\n\n**Impulses** \nManage impulse response files. This menu allows to load a specific impulse or add/remove files from the Easy Effects configuration directory.\n\n**Combine** \nSelect two impulse responses to combine into a single file to be saved under the Easy Effects configuration directory.\n\n**Stereo Width** \nModify the impulse response stereo image width.\n\n**Spectrum** \nVisualize the frequency spectrum of the selected channel.\n\n## References\n\n- [Wikipedia Reverb Effect](https://en.wikipedia.org/wiki/Reverb_effect)\n- [Indiana University - Convolution: A Form of Cross-Synthesis](https://cmtext.indiana.edu/synthesis/chapter4_convolution.php)\n- [Designing Sound - Recording Impulse Responses](https://designingsound.org/2012/12/29/recording-impulse-responses/)\n","src/contents/docs/plugins/crossfeed.md":"# Crossfeed\n\nEasy Effects uses the Crossfeed by bs2b. This plugin is used to improve headphone listening of stereo audio records. It does so by mixing the left and right channel in a way that simulates a stereo speaker setup while using headphones.\n\n**Cutoff** \nLow-Pass filter cutoff frequency.\n\n**Feed** \nAmount of signal from a channel that is sent to the other.\n\n**Presets** \n- **Default** - Closest to virtual speaker placement (30°, 3 meters).\n- **Cmoy** - Close to Chu Moy's Crossfeed (popular).\n- **Jmeier** - Close to Jan Meier's CORDA amplifiers (little change).\n\n## References\n\n- [Wikipedia Crossfeed](https://en.wikipedia.org/wiki/Crossfeed)\n- [Bauer Stereophonic-to-Binaural DSP.](http://bs2b.sourceforge.net/)\n","src/contents/docs/plugins/crosstalkcanceller.md":"# Crosstalk Canceller\n\nWhen audio is played by a stereo speaker audio from both channels arrive to the listener ears. As the superposition\nof each channel audio will happen with a delay that depends on the distance between the listener ear and the speakers\nundesirable effects can happen. That superposition is called crosstalk in this context and can be attenuated by this\nplugin.","src/contents/docs/plugins/crusher.md":"# Crusher\n\nA bitcrusher reduces the resolution or bandwidth of digital audio data. Audio reduced in bit depth sounds more harsh and \"digital\". \nIn the bitcrusher from Calf Studio Gear used in EasyEffects reduction can be done in a linear or logarithmic way.\nAccording to the plugin authors the logarithmic way results in a much smoother sound in low volume signals. \n\n**Mode** \n- **Linear** - Linear distance between bits.\n- **Logarithmic** - Logarithmic distances between bits. The result is a much more \"natural\" sounding crusher which doesn't gate low signals. \n\n**Bit reduction** \nControls the processed audio bit depth. \n\n**DC offset** \nThis offset causes different crushing of the lower and the upper half of the signal. \n\n**Anti-aliasing** \nControls the softness of the crushing sounds. \n\n**Mix** \nControls the mix of original and processed audio. When at 0% the output signal is equal to the original. When at 100% only the processed signal is in the output signal. \n\n**Reduction** \nControls the reduction of the sample rate (downsampling). \n\n**Low frequency oscillator active** \nEnable the low frequency oscillator. The oscillator will make the sample rate reduction change between a range determined by the range control.\n\n**Low frequency oscillator range** \nControls the amount of modulation applied to the sample rate reduction. \n\n**Low frequency oscillator rate** \nControls the frequency of the oscillator. \n\n## References\n\n- [Wikipedia Bit Crusher](https://en.wikipedia.org/wiki/Bitcrusher)\n- [Calf Crusher](https://calf-studio-gear.org)\n- [Calf Crusher Documentation](https://calf-studio-gear.org/doc/Crusher.html)\n","src/contents/docs/plugins/crystalizer.md":"# Crystalizer\n\nThe Crystalizer plugin can be used to add a little of dynamic range to songs that were overly compressed. The signal is split in multiple bands to which different intensities can be applied in order to alter the overall dynamic range.\n\n**Intensities**\nThe higher the value the higher is the difference in magnitude between the loudest and the quietest sounds of the selected band. Different intensities can be set for each frequency band.\n\n**Bypass**\nWhen active the audio signal passing through the selected band is not modified.\n\n**Mute**\nMutes the selected band.\n\n**Adaptive Intensity**\nWhen enabled the intensity value set by the band slider is continuously scaled as audio is processed. The scaling may\nincrease or decrease the intensity based on the geometric mean of the signal crest factor, kurtosis and spectral flux.\n\nWhen using static intensity values it may happen that the crystalizer enhances high frequency audio too much. What leads\nto audible artifacts. The adaptive scaling removes this is most of the cases.\n\n**Oversampling**\nResamples the signal to double its sampling rate value and back to the original value after the plugins effects are applied.\nIt improves the effects quality but it also increases CPU usage.\n\n**Oversampling Quality**\nControls the resampling quality. Higher values require more CPU power.\n\n**Fixed Quantum**\nForces the plugin to use a fixed audio buffer size to process audio. This helps to avoid noises when the sound server switches latency on the fly.\nBut it has the downside of introducing unneeded latency in some cases. And increased CPU usage in cases where higher latency could be used.\n\n**Transition Band**\nControls the width of the transition band of the bandpass filters used to split the signal in several bands. Higher values will use less CPU. But will result in more overlap between the bands.\n\n## References\n\n- [Wikipedia Dynamic Range](https://en.wikipedia.org/wiki/Dynamic_range)\n- [Crest factor](https://en.wikipedia.org/wiki/Crest_factor)\n- [Kurtosis](https://en.wikipedia.org/wiki/Kurtosis)\n- [Spectral flux](https://en.wikipedia.org/wiki/Spectral_flux)\n","src/contents/docs/plugins/deepfilternet.md":"# Deep Noise Remover\n\nAdvanced noise reduction is achieved using \"DeepFilterNet\" technology. This technology uses AI technology called deep learning to efficiently remove noise from audio signals, enabling higher quality noise suppression than conventional methods. This technology has several control parameters, which are explained below and how to set them.\n\n**Model** \nSelects the noise reduction model. Each one must be installed separately.\n\n- DeepFilterNet: requires [DeepFilterNet](https://github.com/Rikorose/DeepFilterNet).\n- DPDFNet: requires [HushMic](https://github.com/Fovty/HushMic). The Attenuation Limit and Process as Mono controls apply to this model.\n\n**Process as Mono (DPDFNet only)**\n\nAverages the left and right input channels before resampling, processes the result with one model instance, and sends the same processed signal to both output channels. This can significantly reduce CPU usage for mono microphones that supply the same signal on both channels. It does not change the selected model or its noise reduction quality.\n\nThis setting is off by default and is saved in presets. Enable it for mono speech input; leave it off when stereo separation matters. If only one input channel contains audio, averaging reduces its level by 6 dB. Opposite-polarity signals can cancel, so check the input channels before enabling it.\n\n**Attenuation Limit** \nThis parameter determines how much noise is attenuated. Higher values result in stronger noise reduction, but too high a value may also remove parts of the audio.\n\n- Recommended setting: 70dB to 80dB is a typical balanced value. This minimizes background noise while preserving speech intelligibility.\n\n**Minimum Processing Threshold** \nSignals below the Minimum Processing Threshold are targeted for denoising. Signals louder than this threshold are not subject to noise reduction and are passed through unchanged. This setting is intended to effectively remove small background noise.\n\n- Recommended setting: -15dB to -30dB is good. This should be fine-tuned according to the ambient noise. Currently, Easy Effects does not allow a value smaller than -15dB.\n\n**Maximum ERB Processing Threshold** \nThe human auditory system has the ability to separate and recognize different frequency components. Equivalent Rectangular Bandwidth (ERB) is a numerical expression and model of this filtering capability. ERB-based processing emphasizes the perceptual importance of speech by focusing on specific frequency bands. This improves the effectiveness of noise reduction and speech enhancement. This threshold sets the maximum level at which perceptually equivalent bandwidth (ERB)-based processing is applied. Signals above this threshold are not subject to noise reduction or speech enhancement processing. This prevents important speech components or loud portions of speech from being unnecessarily processed.\n\n- Recommended setting: Start in the range of 20dB to 30dB and adjust as needed.\n\n**Maximum DF Processing Threshold** \nThe Maximum DF Processing Threshold sets the maximum level of signal that DeepFilterNet will process. Signals exceeding this threshold are excluded from noise reduction processing. This prevents important or loud portions of speech from being unnecessarily processed.\n\n- Recommended setting: 20 dB as the default value. This effectively suppresses noise while maintaining voice quality.\n\n**Minimum Processing Buffer** \nDefines the size of the buffer used to process frames of audio data. A larger buffer can improve noise suppression but may increase latency.\n\n- Recommended setting: Set to a small value (0) if real-time processing is important, or to a number greater than 1 if noise suppression accuracy is a priority.\n\n**Post Filter Beta** \nControls the intensity of the post-processing filter applied after the initial noise suppression. This allows for more subtle refinement of the audio signal.\n\n- Recommended setting: 0.5dB to 2dB is generally recommended, but currently Easy Effects has a maximum value of 0.05dB, which is almost ineffective.\n","src/contents/docs/plugins/deesser.md":"# Deesser\n\nA Deesser is used to dynamically reduce high frequencies. The standard field of use of this plugin is the reduction of \"sssss\" and \"shhhh\" in vocal tracks. Easy Effects uses the Deesser developed by Calf Studio Gear.\n\n**Detection** \nSelect the detection of the Sidechain signal between Peak (stronger) and RMS (smoother).\n\n**Mode** \nSelect the operation mode between Wideband and Split. In Split mode not the full range signal will be affected by the gain reduction, but only frequencies above the split frequency will be manipulated in gain.\n\n**F1 Split** \nThe split frequency. All signals above this frequency will affect the gain reduction (and are affected in Split mode too).\n\n**F1 Gain** \nIt shifts the volume of the higher band. In Wideband mode it affects the Sidechain. In Split mode it also affects the processed high frequencies.\n\n**F2 Peak** \nCenter frequency of the bell filter. It allows a more precise selection of the Sidechain signal.\n\n**F2 Level** \nIncreases or decreases the level of the chosen F2 frequency.\n\n**F2 Peak Q** \nSet the quality of the bell filter. Higher values will affect a narrower frequency range. Lower values will affect a wider band.\n\n**Laxity** \nThe reaction of the Deesser. Higher values won't affect really short peaks.\n\n**Threshold** \nThe level above which the gain reduction is applied.\n\n**Ratio** \nThe amount of attenuation applied to the signal.\n\nFor example, a Ratio of 2 means that if the level rises 4 dB above the Threshold, it will be only 2 dB above after the reduction.\n\n**Makeup** \nThe gain to apply after the processing stage. In Split mode only the high band will be made up.\n\n## References\n\n- [Wikipedia De-essing](https://en.wikipedia.org/wiki/De-essing)\n- [Calf Deesser](https://calf-studio-gear.org/doc/Deesser.html)\n- [LedgerNote - De-esser: The Guide for Sibilant-Free Vocal Recordings](https://ledgernote.com/columns/mixing-mastering/de-esser/)\n","src/contents/docs/plugins/delay.md":"# Delay\n\nThis plugin allows the user to add a short Delay of time to each individual channel of the stereo stream. Easy Effects Delay uses the Delay Compensator Stereo by Linux Studio Plugins set in time mode.\n\n**Left Delay** \nLeft channel delay in milliseconds.\n\n**Right Delay** \nRight channel delay in milliseconds.\n\n**Left Dry Level** \nAmount of right unprocessed signal mixed in the output.\n\n**Right Dry Level** \nAmount of right unprocessed signal mixed in the output.\n\n**Left Wet Level** \nAmount of left processed signal mixed in the output.\n\n**Right Wet Level** \nAmount of right processed signal mixed in the output.\n\n## References\n\n- [Wikipedia Delay (audio effect)](https://en.wikipedia.org/wiki/Delay_(audio_effect))\n- [LSP Delay Compensator Stereo](http://lsp-plug.in/?page=manuals§ion=comp_delay_x2_stereo)\n","src/contents/docs/plugins/echocanceller.md":"# Echo Canceller\n\nThe Echo is a reflected sound wave with sufficient magnitude and delay to be detectable as a signal distinct from the source one. An Echo Canceller is used to improve voice quality by preventing Echo from being created or removing it after it has been added to the source signal. Easy Effects uses the Echo Canceller from SpeexDSP library.\n\n**Frame Size** \nThe amount of time in milliseconds to process at once. It is recommended to use a frame size in the order of 20 ms.\n\n**Filter Length** \nThe amount of time of the Echo cancelling filter to use (also known as tail length). The recommended tail length is approximately the third of the room reverberation time. For example, in a small room, reverberation time is in the order of 300 ms, so a tail length of 100 ms is a good choice.\n\n## References\n\n- [Wikipedia Echo Suppression and Cancellation](https://en.wikipedia.org/wiki/Echo_suppression_and_cancellation)\n- [Speex Acoustic Echo Canceller](https://www.speex.org/docs/manual/speex-manual/node4.html#SECTION00450000000000000000)\n","src/contents/docs/plugins/equalizer.md":"# Equalizer\n\nThe Equalization in sound recording and reproduction is the process of adjusting the volume of different frequency bands within an audio signal. Easy Effects uses the Parametric Equalizer from Linux Studio Plugins. The user can choose from 1 to 32 bands. Width and center frequency of each band can be customized as needed.\n\n## Global Options\n\n**Bands** \nThe number of bands.\n\n**Mode** \n- **IIR** - Infinite Impulse Response filters, nonlinear minimal phase. In most cases does not add noticeable latency to output signal.\n- **FIR** - Finite Impulse Response filters with linear phase, finite approximation of Equalizer's impulse response. Adds noticeable latency to output signal.\n- **FFT** - Fast Fourier Transform approximation of the frequency chart, linear phase. Adds noticeable latency to output signal.\n\n**Balance** \nBalance between left and right output channels.\n\n**Pitch Left** \nThe frequency shift for all filters of the left channel, in semitones.\n\n**Pitch Right** \nThe frequency shift for all filters of the right channel, in semitones.\n\n**Split Channels** \nWhen enabled it is possible to apply different configurations to left and right channels.\n\n**Flat Response** \nThis function sets each band gain to 0.\n\n**Calculate Frequencies** \nThis function calculates the center frequency and the width of each band using the current number of bands. Useful when the user wants fewer than 32 bands but has no idea about which frequencies should be chosen.\n\n## Band Options\n\n**Type** \n- **Off** - The filter is not working (turned off).\n- **Bell** - Bell filter with smooth peak/recess.\n- **High Pass** - High Pass filter with rejection of low frequencies.\n- **High Shelf** - Shelving filter with adjustment of high frequency range.\n- **Low Pass** - Low Pass filter with rejection of high frequencies.\n- **Low Shelf** - Shelving filter with adjustment of low frequency range.\n- **Notch** - Notch filter with full rejection of selected frequency.\n- **Resonance** - Resonance filter with sharp peak/recess.\n- **All Pass** - All Pass filter.\n\n**Mode** \n- **RLC** - Very smooth filters based on similar cascades of RLC contours. Bilinear Z-transform (BT) or Matched Z-transform (MT) is used for pole/zero mapping.\n- **BWC** - Butterworth-Chebyshev-type-1 based filters. Does not affect Resonance and Notch filters. Bilinear Z-transform (BT) or Matched Z-transform (MT) is used for pole/zero mapping.\n- **LRX** - Linkwitz-Riley based filters. Does not affect Resonance and Notch filters. Bilinear Z-transform (BT) or Matched Z-transform (MT) is used for pole/zero mapping.\n- **APO** - Digital biquad filters derived from canonic analog biquad prototypes digitalized through Bilinear transform. These are [textbook filters](https://shepazu.github.io/Audio-EQ-Cookbook/audio-eq-cookbook.html) which are implemented as in the [Equalizer APO](https://equalizerapo.com/) software. Direct design (DR) is used to serve the digital filter coefficients directly in the digital domain, without performing transforms.\n\n**Slope** \nThe slope of the filter characteristics.\n\n**Solo** \nMakes the selected band the only one active.\n\n**Mute** \nMutes the selected band.\n\n**Frequency** \nCenter frequency of the selected band.\n\n**Width** \nBandwidth calculated as `width = frequency / quality`.\n\n**Quality** \nThe quality factor of the filter used.\n\n## References\n\n- [Wikipedia Equalization (audio)](https://en.wikipedia.org/wiki/Equalization_(audio))\n- [LSP Parametric Equalizer x32 LeftRight](http://lsp-plug.in/?page=manuals§ion=para_equalizer_x32_lr)\n- [Wikipedia Q Factor](https://en.wikipedia.org/wiki/Q_factor)\n- [How to EQ - Q Factor and Bandwidth in EQ: What They Mean](https://howtoeq.wordpress.com/2010/10/07/q-factor-and-bandwidth-in-eq-what-it-all-means/)\n","src/contents/docs/plugins/exciter.md":"# Exciter\n\nAn Exciter is used to produce high sound that is not present in the original signal. This is done by creating harmonic distortions of the signal which are restricted in range and added to the original signal. It raises the upper end of an audio signal without simply raising the higher frequencies like an [Equalizer](equalizer.md) would do to create a more \"crisp\" or \"brilliant\" sound.\n\n**Amount** \nAmount of harmonics added to the original signal.\n\n**Harmonics** \nAmount of newly created harmonics.\n\n**Scope** \nThe frequency above which harmonics are produced.\n\n**Ceiling (button)** \nConstrain the excitement on the upper end.\n\n**Ceiling (value)** \nThe frequency above which no harmonics are produced.\n\n**Blend Harmonics** \nThe \"colour\" (or octave) of the harmonics.\n\n**Listen** \nMute the original signal and listen to the harmonics exclusively.\n\n## References\n\n- [Calf Exciter](https://calf-studio-gear.org/doc/Exciter.html)\n- [Wikipedia Exciter (effect)](https://en.wikipedia.org/wiki/Exciter_(effect))\n","src/contents/docs/plugins/expander.md":"# Expander\n\nAn expander is an effect that works inversely to a compressor, further reducing the volume of quiet sounds below a certain level. This allows for natural suppression of background noise and unwanted whispers. For example, it is effective when air conditioning or other environmental sounds intrude into quiet scenes.\n\nThis effect is similar to the gate, which is explained in another chapter, in that it blocks quiet sounds. However, while a gate completely cuts sound, an expander gradually reduces sound, resulting in a more natural and smooth decay characteristic. Its main uses are listed below:\n\n- Reduce the prominence of quiet environmental noises (air conditioning, PC fans, clicks, etc.) during periods of silence\n- Suppress noise in quiet sections while maintaining the clarity of narration and dialogue\n- Emphasize the contours (punch) of sounds to create a more dynamic impression\n\n## Expander Options\n\nThe following is an explanation of each parameter when the mode is set to \"**Downward**.\"\n\n**Attack Time**: This determines the time it takes for the expander to begin suppressing the sound after the sound volume drops below the threshold. Shorter values will reduce the volume immediately, quickly suppressing noise and other imperfections. Longer values will cause the processing to begin later, preserving the natural reverberation and nuances of soft sounds.\n\n**Release Time**: This determines the time it takes for the expander to stop suppressing the volume and return to its original volume after the sound exceeds the threshold again and is determined to no longer require processing.\n\nLonger values will cause the sound to return more slowly, making it sound more natural.\n\nShorter values will return the sound to its original volume quickly, making it sound clearer and tighter, but may sound unnatural in some cases.\n\n**Attack Threshold**: The effect is only applied to sounds quieter than this value.\n\n**Release Threshold**: This can be the same as the Attack Threshold, but setting it about -5dB lower than the Attack Threshold will enable the Hysteresis function, suppressing fluttering around the threshold and achieving more stable processing.\n\n**Ratio**: This is an indicator of how much quieter (= how much) sounds below the threshold will be reduced. Set the value using the following as a guide:\n\n- A setting of 1 passes the signal as is (same as Expander OFF)\n- For light noise suppression (natural processing), use 1.5-3\n- For clarity of dialogue and smoothing out quiet sections, use 4-8\n- For complete noise reduction (gate-like behavior), use 20-100\n\n*Please note that setting the ratio too high can result in a choppy, unnatural sound.\n\n**Knee**: This parameter adjusts how smoothly processing begins near the threshold. The name comes from the fact that the curve on the graph resembles a knee. See below for the relationship between the setting value and the curve.\n\nKnee Value Volume Curve Characteristics Processing Sensation\n\n0dB Abrupt bend (hard knee) The effect is applied suddenly (like an ON/OFF switch)\n\n-6dB Gradual bend Processing begins naturally\n\n-12dB and above Very smooth Processing is barely noticeable\n\n**Makeup**: The gain applied after gating.\n\n**Dry Level**: The amount of unprocessed signal mixed with the output.\n\n**Wet Level**: The amount of processed signal mixed with the output.\n\nFor information on how to set up sidechaining and sidechain filtering, see the chapter of Gate.\n\n*When \"**Upward**\" is selected in the mode setting, the expander will function to boost (increase the volume of) quieter sounds.\n\nThis can be used in situations where dialogue or narration is too quiet to be heard, and is also effective when you want to bring out the subtle nuances of a musical performance (such as pianissimo).\n\nRather than \"lowering louder sounds\" like a compressor, it is used to increase the average volume by \"boosting quieter sounds,\" or to increase sound pressure while maintaining the punch and three-dimensionality of the original sound.","src/contents/docs/plugins/filter.md":"# Filter\n\nA Filter is used to amplify (boost), pass or attenuate (cut) defined parts of a frequency spectrum. Easy Effects uses the Filter from Linux Studio Plugins.\n\n**Type** \n- **Low-pass** - Low-pass filter with rejection of high frequencies. \n- **High-pass** - High-pass filter with rejection of low frequencies. \n- **Low-shelf** - Shelving filter with adjustment of low frequencies. \n- **High-shelf** - Shelving filter with adjustment of high frequency range. \n- **Bell** - Bell filter with smooth peak/recess. \n- **Bandpass** - Bandpass filter. \n- **Notch** - Notch filter with full rejection of selected frequency. \n- **Resonance** - Resonance filter with sharp peak/recess. \n- **Ladder-pass** - The filter that makes some ladder-passing in the spectrum domain. \n- **Ladder-rej** - The filter that makes some ladder-rejection in the spectrum domain. \n- **Allpass** - All-pass filter which only affects the phase of the audio signal at the specified frequency. \n\n**Filter Mode** \n- **RLC** - Very smooth filters based on similar cascades of RLC contours. \n- **BWC** - Butterworth-Chebyshev-type-1 based filters. Does not affect Resonance and Notch filters. \n- **LRX** - Linkwitz-Riley based filters. Does not affect Resonance and Notch filters. \n- **APO** - Digital biquad filters derived from canonic analog biquad prototypes digitalized through Bilinear transform. These are textbook filters which are implemented as in the EqualizerAPO software. \n- **BT** - Bilinear Z-transform is used for pole/zero mapping. \n- **MT** - Matched Z-transform is used for pole/zero mapping. \n- **DR** - Direct design is used to serve the digital filter coefficients directly in the digital domain, without performing transforms. \n\n**Equalizer Mode** \n- **IIR** - Infinite Impulse Response filters, nonlinear minimal phase. In most cases does not add noticeable latency to output signal. \n- **FIR** - Finite Impulse Response filters with linear phase, finite approximation of equalizer's impulse response. Adds noticeable latency to output signal. \n- **FFT** - Fast Fourier Transform approximation of the frequency chart, linear phase. Adds noticeable latency to output signal. \n- **SPM** - Spectral Processor Mode of equalizer, equalizer transforms the magnitude of signal spectrum instead of applying impulse response to the signal.\n\n**Slope** \nThe slope of the filter characteristics. \n\n**Frequency** \nThe cutoff/resonance frequency of the filter or the middle frequency of the band. \n\n**Width** \nThe width of the bandpass/ladder filters in octaves. \n\n**Gain** \nThe gain of the filter. It is disabled for lo-pass/hi-pass/notch filters. \n\n**Quality** \nThe quality factor of the filter. \n\n**Balance** \nThe balance between left and right output channels. \n\n## References\n\n- [Wikipedia Audio Filter](https://en.wikipedia.org/wiki/Audio_filter)\n- [Linux Studio Plugins Filter](https://lsp-plug.in/?page=manuals§ion=filter_stereo)\n","src/contents/docs/plugins/gate.md":"# Gate\n\nThe Gate attenuates signals that register below a Threshold. This kind of signal processing is used to reduce disturbing noise between useful signals. Easy Effects uses the Stereo Sidechain Gate from Linux Studio Plugins.\n\n## Gate Workflow\n\nThe Gate begins to open when the Sidechain level becomes above the Attack Zone Start level.\n\nThe Gate fully opens when the Sidechain level becomes above the Attack Threshold level.\n\nThe Gate begins to close when the Sidechain level becomes below the Release Zone Start level.\n\nThe Gate fully closes when the Sidechain level becomes below the Release Threshold level.\n\n## Gate Options\n\n**Attack Time** \nThe length of time it takes to restore roughly two-thirds of the gain reduction.\n\n**Release Time** \nThe length of time it takes to apply roughly two-thirds of the gain reduction.\n\n**Curve Threshold** \nThe Gate fully opens upon the Sidechain level becoming above Curve Threshold (displayed as Attack Threshold level).\n\nIf Hysteresis is not enabled, the Gate begins to close upon the Sidechain level becoming below Curve Threshold (displayed as Release Zone Start level).\n\n**Curve Zone Size** \nThe Gate begins to open upon the Sidechain level becoming above the Curve Threshold + Curve Zone (displayed as Attack Zone Start level).\n\nIf Hysteresis is not enabled, the Gate fully closes upon the Sidechain level becoming below the Curve Threshold + Curve Zone (displayed as Release Threshold level).\n\n**Hysteresis** \nWhen enabled, Curve Threshold and Curve Zone apply only to the opening Gate, and separate parameters can be configured for closing Gate.\n\n**Hysteresis Threshold** \nIf Hysteresis is enabled, the Gate begins to close upon the Sidechain level becoming below Curve Threshold + Hysteresis Threshold (displayed as Release Zone Start level).\n\n**Hysteresis Zone Size** \nIf Hysteresis is enabled, the Gate fully closes upon the Sidechain level becoming below the Curve Threshold + Hysteresis Threshold + Hysteresis Zone (displayed as Release Threshold level).\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal mixed in the output.\n\n**Reduction** \nIf the value is negative, it acts as the amount of gain reduction to apply to the input signal when the Gate is fully closed. If the value is positive, the Gate operates in \"Reverse Mode\": It reduces the amplitude when the signal is above the threshold.\n\n**Makeup** \nThe gain to apply after the gating stage.\n\n## Sidechain\n\n**Listen** \nAllows to listen the processed Sidechain signal.\n\n**Input Type** \nDetermines which signal is the Sidechain or, in other words, the signal that controls the gating stage.\n\n- **Internal** - The Sidechain is the Gate input signal (taken after applying the plugin input gain).\n- **External** - The Sidechain is an external source took by a specific input device (typically a microphone).\n\n**Input Device** \nSelect the device for the External Sidechain.\n\n**Mode** \nDetermines how the Sidechain is evaluated for the gating stage.\n\n- **Peak** - The Gate reacts according to the peaks.\n- **RMS** - The Gate reacts according to the average loudness measured by the root mean square.\n- **Low-Pass Filter** - The Gate reacts according to the signal processed by a recursive 1-pole Low-Pass filter.\n- **Simple Moving Average** - The Gate reacts according to the signal processed by the Simple Moving Average filter.\n\n**Source** \nDetermines which part of the Sidechain is taken into account for the gating stage.\n\n- **Middle** - The sum of left and right channels.\n- **Side** - The difference between left and right channels.\n- **Left** - Only left channel is used.\n- **Right** - Only right channel is used.\n- **Min** - The absolute minimum value is taken from stereo input.\n- **Max** - The absolute maximum value is taken from stereo input.\n\n**PreAmplification** \nGain applied to the Sidechain signal.\n\n**Reactivity** \nThe time that defines the number of samples used to process the Sidechain in RMS, Uniform and Low-Pass modes. Higher the value, more smooth the gating.\n\n**Lookahead** \nThe signal to gate is delayed by this amount of time, so that the gating will be applied earlier than it would be otherwise. The corresponding delay is reproduced on the output signal.\n\n## Sidechain Filters\n\n**High-Pass Filter Mode** \nSets the type of the High-Pass filter applied to Sidechain signal.\n\n**High-Pass Frequency** \nSets the cut-off frequency of the High-Pass filter.\n\n**Low-Pass Filter Mode** \nSets the type of the Low-Pass filter applied to Sidechain signal.\n\n**Low-Pass Frequency** \nSets the cut-off frequency of the Low-Pass filter.\n\n## References\n\n- [Wikipedia Dynamic Range Compression](https://en.wikipedia.org/wiki/Dynamic_range_compression)\n- [LSP Sidechain Gate Stereo](https://lsp-plug.in/?page=manuals§ion=sc_gate_stereo)\n- [Wikipedia Noise Gate](https://en.wikipedia.org/wiki/Noise_gate)\n","src/contents/docs/plugins/harmonizer.md":"# Harmonizer\n\nThe Harmonizer adds two pitch shifted voices to the input. Set the root note and chord, then choose an interval and level for each voice. The dry/wet control balances the original signal with the harmonized signal.\n\nEasy Effects uses the StereoHarm LV2 plugin from Rakarrack Plus. Install Rakarrack Plus to use this effect.\n\n## References\n\n- [Rakarrack Plus](https://github.com/Stazed/rakarrack-plus)\n- [Harmonizer documentation](https://github.com/Stazed/rakarrack-plus)\n","src/contents/docs/plugins/limiter.md":"# Limiter\n\nA Limiter is a special type of downward [Compressor](compressor.md) which does not allow the signal to overtake a predetermined Threshold. Ideally it has a very high compression ratio that takes the amplitude below a ceiling which stands as the maximum output level. For this reason it is usually named \"brick-wall limiter\".\n\nEasy Effects uses the Sidechain Stereo Limiter from Linux Studio Plugins. In most cases it works as a brick-wall limiter, but it offers also an additional feature that acts like a Compressor with extreme settings, so the output signal may exceed the specified Threshold.\n\n**Mode** \nSelect the operative mode of the peak cutting algorithm which searches the peaks above the Threshold and applies short gain reduction patches to the signal.\n\nThese patches can be selected in 3 forms: **Hermite**, **Exponential** and **Linear**. Each one has 4 different variants related to gain reduction of the samples around the peak: **Thin**, **Tail**, **Duck** and **Wide**.\n\nThe shape of each form and variant can be referred into the Linux Studio Plugin manual. See References section at the bottom of the present document.\n\n**Oversampling** \nWhen enabled, the sample rate is internally increased in order to improve peak detection and reduce aliasing (that causes distortion).\n\nThe modes have 2 main types: **Full**, which increases both the Sidechain and the Input signals, and **Half** which increases only the Sidechain. Each one of them specifies different **multipliers** and, between parentheses, the number of \"**lobes** in the kernel\".\n\nThe oversampled Input signal is downsampled to the original sample rate after processing.\n\n**Dither** \nIf specified, it enables Dithering for the selected bit depth. The process of Dithering adds a low-level noise to output signal in order to mask \"quantization distortion\", a form of artifact present in digital audio rendered at lower bit depth.\n\n**SC PreAmp** \nThe gain applied to the Sidechain before it is processed.\n\n**Lookahead** \nThe size of the buffer used to detect the peaks in advance. It adds the corresponding latency to the output signal.\n\n**Attack** \nThe length of time it takes to apply the needed gain reduction to keep the peak below the Threshold.\n\nIt affects the length of the gain reduction patch. It cannot be larger than the Lookahead (if specified larger, it's set internally as the max possible value).\n\n**Release** \nThe length of time it takes to restore the reduced gain around the limited peak.\n\nIt affects the length of the gain reduction patch. It cannot be twice larger than the Lookahead (if specified larger, it's set internally as the max possible value).\n\n**Threshold** \nThe target level above which the Limiter should reduce the peak of the signal. In some modes it represents the maximum output level.\n\n**Threshold Boost** \nIf enabled it applies an amount of gain to the limited signal equal to the absolute value of the Threshold. This causes the peak limited at the Threshold level to output at 0 dB (it has only effect when the Threshold is set below 0 dB).\n\n**Stereo Link** \nThe degree of the channel linking. At 0% both channels are limited independently while at 100% the loudest one triggers the same gain reduction on both.\n\n**External Sidechain** \nSwitch the Sidechain to an external source took by a specific input device (typically a microphone).\n\n## Auto Leveling\n\nThe Auto Leveling checkbutton introduces an additional feature named \"Automated Level Regulation\" (ALR) which acts like a Compressor with infinite ratio for the purpose of applying a smoothed gain reduction rather than a stronger peak cutter like in brick-wall mode. This configuration could get the output level to exceed the Threshold even if the signal is highly compressed.\n\n**Attack** \nManage how the raise of the input signal affects the smoothness of the ALR curve that controls the gain reduction level. Higher the value, more quickly the curve goes to it's maximum.\n\n**Release** \nManage how the fall of the input signal affects the smoothness of the curve that controls the gain reduction level. Higher the value, more quickly the curve goes to it's minimum.\n\n**Knee** \nManage the Threshold of the ALR gain curve and, in fact, adjust the balance between two gain reduction stages. Raising the value delegates more work to the peak-cutting algorithm. Lowering the value delegates more work to the ALR gain reduction algorithm.\n\n## References\n\n- [Wikipedia Limiter](https://en.wikipedia.org/wiki/Limiter)\n- [LSP Sidechain Limiter Stereo](https://lsp-plug.in/?page=manuals§ion=sc_limiter_stereo)\n","src/contents/docs/plugins/loudness.md":"# Loudness\n\nEasy Effects uses the Loudness Compensator from Linux Studio Plugins which applies the \"equal-loudness contour\" corrections to the input signal.\n\nAn equal-loudness contour is a measure of sound pressure level (SPL), over the frequency spectrum, for which a listener perceives a constant loudness when presented with pure steady tones. The unit of measurement for loudness levels is the phon and is arrived at by reference to equal-loudness contours.\n\nUsage of equal-loudness contours solves many mixing problems that every sound engineer meets while working on the track. The main problem is that human ear perceives different frequencies for different volume settings in a different way. In other words, applying changes to the mix on the low volume settings may cause unexpected sounding of the mix at the maximum loudness.\n\nThe Loudness Compensator performs frequency response computations and applies the computed frequency response to the input signal depending on the output volume settings. Additionally it can provide ear protection by applying hard-clipping to the output signal if it exceeds the allowed configurable level.\n\n**Standard** \nAllows to select different equal-loudness contour standards.\n\n- **Flat** - Applies flat frequency response to the whole spectrum. It's similar to just a gain knob but useful to perform a comparison to other modes.\n- **ISO 226:2003** - Recent equal-loudness contour standard published in 2003.\n- **Fletcher-Munson** - The first equal-loudness contour implementation by Harvey Fletcher and Wilden A. Munson published in 1933.\n- **Robinson-Dadson** - More accurate equal-loudness contour implementation by D.W. Robinson and R.S. Dadson published in 1956. It became the basis for the ISO 226:2003 standard.\n\n**FFT Size** \nAllows to select size of the Fast Fourier Transform frame used for the processing. The larger FFT frame is, the more precise the curve is approximated and the more latency is introduced.\n\n**Output Volume** \nThe output volume of the signal with applied equal loudness contour. It controls the loudness of the 1 kHz pure sine wave.\n\n**Clipping** \nAllows to enable and the hard clipping of the output signal.\n\n**Clipping Range** \nAllows to set the gap level for the hard clipping of the output signal.\n\n## References\n\n- [Wikipedia Equal-Loudness Contour](https://en.wikipedia.org/wiki/Equal-loudness_contour)\n- [LSP Loudness Compensator Stereo](https://lsp-plug.in/?page=manuals§ion=loud_comp_stereo)\n- [Lindos Electronics - Equal-Loudness Contours](http://www.lindos.co.uk/cgi-bin/FlexiData.cgi?SOURCE=Articles&VIEW=full&id=17)\n","src/contents/docs/plugins/maximizer.md":"# Maximizer\n\nThe Maximizer is a special type of [Limiter](limiter.md) that does not only prevent the signal to exceed a specified target level, but also adjusts the average Loudness of the audio track.\n\nEasy Effects uses the Maximizer developed by ZamAudio. It acts like an Amplifier that feeds a brick-wall Limiter with a fixed Lookahead of 10 ms adding the corresponding delay to the output signal.\n\n**Threshold** \nThis parameter represents the ideal amplification level needed by the signal.\n\nIt contributes along with the Ceiling to determine the amount of gain to apply to the signal before the limiting stage. The gain is calculated by `(-Threshold) - (-Ceiling)`.\n\nFor example, on Threshold -6 dB and Ceiling -2 dB, the signal is amplified by 4 dB and limited to not exceed -2 dB output level.\n\n**Ceiling** \nThis parameter represents the ideal attenuation level needed by the signal and the maximum allowed output level.\n\nWhen the Threshold is set to 0 dB, the Ceiling is simply the gain reduction. When the Threshold is lowered, the signal is boosted without overtaking the Ceiling value.\n\n**Release** \nSets the release of the internal brick-wall Limiter. Lower values may introduce small artifacts.\n\n## References\n\n- [Sonarworks Blog - What Is A Maximizer?](https://www.sonarworks.com/soundid-reference/blog/learn/what-is-a-maximizer/)\n","src/contents/docs/plugins/midside_equalizer.md":"# Mid-Side Equalizer\n\nThe Mid-Side Equalizer processes the Mid (sum) and Side (difference) parts of a stereo signal independently. It's useful, for example, to fix the overall tone of a recording through the Mid channel while keeping the stereo ambience untouched on the Side channel. Easy Effects uses the Parametric Equalizer x32 MidSide from Linux Studio Plugins. You can run anywhere from 1 to 32 bands per channel, and tweak the width and center frequency of each band.\n\n## Global Options\n\n**Bands** \nNumber of bands.\n\n**Mode** \n- **IIR** - Infinite Impulse Response filters, nonlinear minimal phase. Usually doesn't add noticeable latency to the output.\n- **FIR** - Finite Impulse Response filters with linear phase, a finite approximation of the equalizer's impulse response. Adds noticeable latency to the output.\n- **FFT** - Fast Fourier Transform approximation of the frequency chart, linear phase. Adds noticeable latency to the output.\n\n**Balance** \nBalance between the left and right output channels.\n\n**Pitch Mid** \nFrequency shift for all mid channel filters, in semitones.\n\n**Pitch Side** \nFrequency shift for all side channel filters, in semitones.\n\n**Link Mid/Side** \nWhen enabled, you can apply different configurations to the mid and side channels.\n\n**Flat Response** \nResets all band gains to 0.\n\n**Calculate Frequencies** \nFigures out the center frequency and width for each band based on the current band count. Handy if you want fewer than 32 bands but aren't sure which frequencies to pick.\n\n## Band Options\n\n**Type** \n- **Off** - Filter is disabled.\n- **Bell** - Bell filter with smooth peak/recess.\n- **High Pass** - High Pass filter that removes low frequencies.\n- **High Shelf** - Shelving filter that adjusts the high frequency range.\n- **Low Pass** - Low Pass filter that removes high frequencies.\n- **Low Shelf** - Shelving filter that adjusts the low frequency range.\n- **Notch** - Notch filter that fully rejects the selected frequency.\n- **Resonance** - Resonance filter with sharp peak/recess.\n- **All Pass** - All Pass filter.\n\n**Mode** \n- **RLC** - Very smooth filters built from cascaded RLC contours. Uses Bilinear Z-transform (BT) or Matched Z-transform (MT) for pole/zero mapping.\n- **BWC** - Butterworth-Chebyshev-type-1 based filters. Doesn't affect Resonance and Notch filters. Uses Bilinear Z-transform (BT) or Matched Z-transform (MT) for pole/zero mapping.\n- **LRX** - Linkwitz-Riley based filters. Doesn't affect Resonance and Notch filters. Uses Bilinear Z-transform (BT) or Matched Z-transform (MT) for pole/zero mapping.\n- **APO** - Digital biquad filters derived from canonical analog biquad prototypes digitalized through Bilinear transform. These are [textbook filters](https://shepazu.github.io/Audio-EQ-Cookbook/audio-eq-cookbook.html) implemented the same way as in [Equalizer APO](https://equalizerapo.com/). Direct design (DR) generates the digital filter coefficients directly in the digital domain, without any transforms.\n\n**Slope** \nSlope of the filter curve.\n\n**Solo** \nMakes the selected band the only active one.\n\n**Mute** \nMutes the selected band.\n\n**Frequency** \nCenter frequency of the band.\n\n**Width** \nBandwidth, calculated as `width = frequency / quality`.\n\n**Quality** \nQuality factor of the filter.\n\n## References\n\n- [Wikipedia Equalization (audio)](https://en.wikipedia.org/wiki/Equalization_(audio))\n- [LSP Parametric Equalizer x32 MidSide](http://lsp-plug.in/?page=manuals§ion=para_equalizer_x32_ms)\n- [Wikipedia Q Factor](https://en.wikipedia.org/wiki/Q_factor)\n- [How to EQ - Q Factor and Bandwidth in EQ: What They Mean](https://howtoeq.wordpress.com/2010/10/07/q-factor-and-bandwidth-in-eq-what-it-all-means/)\n","src/contents/docs/plugins/multibandcompressor.md":"# Multiband Compressor\n\nEasy Effects uses the Sidechain Multiband Compressor Stereo developed by Linux Studio Plugins. Please refer to the [Compressor](compressor.md) documentation to comprehend the basic functionality of the dynamic range compression.\n\n## Global Options\n\n**Band Management** \nThis Compressor allows to split the input signal up to 8 bands. Each band is not attached to its strict frequency and can be controlled by completely different frequency range that can be obtained by applying Low-Cut and Hi-Cut filters to the Sidechain signal.\n\nThe first band is always enabled while the others can be activated if needed. When only the first band is enabled, the functionality is similar to the Singleband Sidechain Compressor. Otherwise the signal is split in more bands and each band is compressed individually. After the compression stage, the bands are mixed together to form the output result.\n\n**Operating Mode** \nDetermines how the input signal is split to obtain the different bands.\n\n- **Classic** - The original signal is split using crossover filters. After the compression stage, all bands become phase-compensated using All-Pass filters.\n- **Modern** - Each band is processed by a pair of dynamic shelving filters. This mode allows a better control over the gain of each band.\n\n**Sidechain Boost** \nIf enabled, it introduces a special mode for assigning the same weight for higher frequencies opposite to lower frequencies.\n\nWhen disabled, the frequency band is processed 'as is', but the usual audio signal has 3 dB/octave falloff in the frequency domain and could be compared with the pink noise. So lower frequencies take more effect on the Compressor rather than higher frequencies.\n\nOn the other hand the Sidechain Boost mode allows to compensate the -3 dB/octave falloff of the signal spectrum and, even more, makes the signal spectrum growing +3 dB/octave in the almost fully audible frequency range.\n\n**Pink** applies +3 dB/octave while **Brown** applies +6 dB/octave Sidechain Boost. Each of them can use bilinear-transformed (BT) or matched-transformed (MT) shelving filter.\n\n**Sidechain Source** \nIf the External Sidechain is enabled inside at least one band, this combobox allows to select the input device as source.\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal mixed in the output.\n\n## Band Options\n\n**Band Start** \nAllows to change the lower end split frequency of the selected band. This value is assigned to the Band End of the previous enabled band. It can be modified for all bands except the first one, which is always enabled and its value is 0 Hz.\n\n**Band End** \nSpecify the upper end split frequency of the selected band. It cannot be directly modified and assumes the same value of the Band Start of the next enabled band. For the last enabled band it is always 24.000 Hz.\n\n**Compression Mode** \n- **Downward** - It's aimed to decrease the gain of the signal above the Threshold.\n- **Upward** - It's aimed to increase the gain of the signal below the Threshold.\n- **Boosting** - It's aimed to increase the gain of the signal below the Threshold by a specific amount.\n\n**External Sidechain** \nThe Sidechain is an external source took by a specific input device (typically a microphone).\n\n**Band Bypass** \nIf enabled, the selected band is not affected by the compression stage.\n\n**Solo** \nTurns on the Solo mode to the selected band by applying -36 dB gain to the other non-soloing bands.\n\n**Mute** \nTurns on the Mute mode applying -36 dB gain to to the selected band.\n\n**Attack Time** \nThe length of time it takes to apply roughly two-thirds of the targeted amount of compression ratio to the uncompressed band signal.\n\n**Release Time** \nThe length of time it takes to restore roughly two-thirds of the reduced gain (in Downward mode) or increased gain (in Upward/Boosting mode) to the compressed band signal.\n\n**Attack Threshold** \nThe target level around which the compression is applied (the range depends by the Knee).\n\n**Release Threshold** \nSets up the Threshold of the Release Time, calculated by summing the Release Threshold to the Attack Threshold.\n\nIf the Sidechain level is above that Threshold, the compressor uses the Release Time for the releasing stage. Otherwise the Attack Time is used in place of Release Time.\n\nFor example, with -10 dB Attack Threshold and -60 dB Release Threshold, if the Sidechain is above `-10 + (-60) = -70 dB`, the Release Time is used for the gain restoration. If the Sidechain is below -70 dB, the Attack Time is used for the gain restoration.\n\n**Ratio** \nThe amount of attenuation (in Downward mode) or amplification (in Upward/Boosting mode) that will be applied to the signal.\n\nFor example, when the Ratio is 2 in Downward mode and the Sidechain rises above the Threshold by 10 dB, the signal would be ideally reduced by 5 dB (10/2 dB). In practice this behavior mostly depends on how the Multiband Compressor is designed and configured.\n\n**Knee** \nThe range over which the Compressor switches from no compression to almost the full ratio compression (the Threshold usually sits at the center of this transition zone).\n\n**Makeup** \nThe gain to apply after the compression stage.\n\n## Band Sidechain Options\n\n**Mode** \nDetermines how the Sidechain of the selected band is evaluated for the compression stage.\n\n- **Peak** - The Compressor reacts according to the peaks.\n- **RMS** - The Compressor reacts according to the average loudness measured by the root mean square.\n- **Low-Pass** - The Compressor reacts according to the signal processed by a Low-Pass filter.\n- **Uniform** - The Compressor reacts according to the loudness measured by the average of the absolute amplitude.\n\n**Source** \nDetermines which part of the Sidechain is taken into account for the compression stage.\n\n- **Middle** - The sum of left and right channels.\n- **Side** - The difference between left and right channels.\n- **Left** - Only left channel is used.\n- **Right** - Only right channel is used.\n- **Min** - The absolute minimum value is taken from stereo input.\n- **Max** - The absolute maximum value is taken from stereo input.\n\n**Low-Cut Filter** \nEnables a custom Low-Cut Filter for the selected band.\n\n**Low-Cut Frequency** \nSets the cut-off frequency of the custom Low-Cut filter. If it is disabled, the default Low-Cut filter assumes internally the Band Start frequency as the cut-off frequency.\n\n**Hight-Cut Filter** \nEnables a custom High-Cut Filter for the selected band.\n\n**Hight-Cut Frequency** \nSets the cut-off frequency of the custom High-Cut filter. If it is disabled, the default High-Cut filter assumes internally the Band End frequency as the cut-off frequency.\n\n**PreAmp** \nGain applied to the Sidechain signal of the selected band.\n\n**Reactivity** \nThe time that defines the number of samples used to process the Sidechain in RMS, Uniform and Low-Pass modes. Higher the value, more smooth the compression.\n\n**Lookahead** \nThe band signal to compress is delayed by this amount of time, so that the compression will be applied earlier than it would be otherwise.\n\nEach band can have different Lookahead values. To avoid phase distortions in the mixing stage, all the bands are automatically delayed for an individually calculated period of time.\n\n**Boost Amount** \nMaximum gain amplification to apply in Boosting Mode.\n\n**Boost Threshold** \nThe Threshold below which a constant amplification will be applied to the band signal in Upward Mode (it prevents from applying infinite amplification to very quiet signals).\n\n## References\n\n- [Wikipedia Dynamic Range Compression](https://en.wikipedia.org/wiki/Dynamic_range_compression)\n- [LSP Sidechain Multiband Compressor Stereo](https://lsp-plug.in/?page=manuals§ion=sc_mb_compressor_stereo)\n- [Black Ghost Audio - The Ultimate Guide to Compression](https://www.blackghostaudio.com/blog/the-ultimate-guide-to-compression)\n- [Attack Magazine - Demolishing The Myths of Compression](https://www.attackmagazine.com/features/columns/gregory-scott-demolishing-the-myths-of-compression/)\n","src/contents/docs/plugins/multibandgate.md":"# Multiband Gate\n\nEasy Effects uses the Multiband Gate Stereo developed by Calf Studio Gear. Please refer to the [Gate](gate.md) documentation to comprehend the basic functionality of the gating process.\n\n## Global Options\n\n**Band Management** \nThis Gate allows to split the input signal up to 8 bands. Each band is not attached to its strict frequency and can be controlled by completely different frequency range that can be obtained by applying Low-Cut and Hi-Cut filters to the Sidechain signal.\n\nThe first band is always enabled while the others can be activated if needed. When only the first band is enabled, the functionality is similar to the Singleband Sidechain Gate. Otherwise the signal is split in more bands and each band is processed individually. After the gating stage, the bands are mixed together to form the output result.\n\n**Operating Mode** \nDetermines how the input signal is split to obtain the different bands.\n\n- **Classic** - The original signal is split using crossover filters. After the gating stage, all bands become phase-compensated using All-Pass filters.\n- **Modern** - Each band is processed by a pair of dynamic shelving filters. This mode allows a better control over the gain of each band.\n\n**Sidechain Boost** \nIf enabled, it introduces a special mode for assigning the same weight for higher frequencies opposite to lower frequencies.\n\nWhen disabled, the frequency band is processed 'as is', but the usual audio signal has 3 dB/octave falloff in the frequency domain and could be compared with the pink noise. So lower frequencies take more effect on the Gate rather than higher frequencies.\n\nOn the other hand the Sidechain Boost mode allows to compensate the -3 dB/octave falloff of the signal spectrum and, even more, makes the signal spectrum growing +3 dB/octave in the almost fully audible frequency range.\n\n**Pink** applies +3 dB/octave while **Brown** applies +6 dB/octave Sidechain Boost. Each of them can use bilinear-transformed (BT) or matched-transformed (MT) shelving filter.\n\n**Sidechain Source** \nIf the External Sidechain is enabled inside at least one band, this combobox allows to select the input device as source.\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal mixed in the output.\n\n## Band Options\n\n**Band Start** \nAllows to change the lower end split frequency of the selected band. This value is assigned to the Band End of the previous enabled band. It can be modified for all bands except the first one, which is always enabled and its value is 0 Hz.\n\n**Band End** \nSpecify the upper end split frequency of the selected band. It cannot be directly modified and assumes the same value of the Band Start of the next enabled band. For the last enabled band it is always 24.000 Hz.\n\n**External Sidechain** \nThe Sidechain is an external source took by a specific input device (typically a microphone).\n\n**Band Bypass** \nIf enabled, the selected band is not affected by the gating stage.\n\n**Solo** \nTurns on the Solo mode to the selected band by applying -36 dB gain to the other non-soloing bands.\n\n**Mute** \nTurns on the Mute mode applying -36 dB gain to to the selected band.\n\n**Attack Time** \nThe length of time it takes to apply roughly two-thirds of the gain reduction.\n\n**Release Time** \nThe length of time it takes to restore roughly two-thirds of the gain reduction.\n\n**Curve Threshold** \nThe Gate fully opens upon the Sidechain level becoming above Curve Threshold.\n\nIf Hysteresis is not enabled, the Gate begins to close upon the Sidechain level becoming below Curve Threshold.\n\n**Curve Zone Size** \nThe Gate begins to open upon the Sidechain level becoming above the Curve Threshold + Curve Zone.\n\nIf Hysteresis is not enabled, the Gate fully closes upon the Sidechain level becoming below the Curve Threshold + Curve Zone.\n\n**Hysteresis** \nWhen enabled, Curve Threshold and Curve Zone apply only to the opening Gate, and separate parameters can be configured for closing Gate.\n\n**Hysteresis Threshold** \nIf Hysteresis is enabled, the Gate begins to close upon the Sidechain level becoming below Curve Threshold + Hysteresis Threshold.\n\n**Hysteresis Zone Size** \nIf Hysteresis is enabled, the Gate fully closes upon the Sidechain level becoming below the Curve Threshold + Hysteresis Threshold + Hysteresis Zone.\n\n**Reduction** \nIf the value is negative, it acts as the amount of gain reduction to apply to the input signal when the Gate is fully closed. If the value is positive, the Gate operates in \"Reverse Mode\": It reduces the amplitude when the signal is above the threshold.\n\n**Makeup** \nThe gain to apply after the gating stage.\n\n## Band Sidechain Options\n\n**Mode** \nDetermines how the Sidechain is evaluated for the gating stage.\n\n- **Peak** - The Gate reacts according to the peaks.\n- **RMS** - The Gate reacts according to the average loudness measured by the root mean square.\n- **Low-Pass Filter** - The Gate reacts according to the signal processed by a recursive 1-pole Low-Pass filter.\n- **Simple Moving Average** - The Gate reacts according to the signal processed by the Simple Moving Average filter.\n\n**Source** \nDetermines which part of the Sidechain is taken into account for the gating stage.\n\n- **Middle** - The sum of left and right channels.\n- **Side** - The difference between left and right channels.\n- **Left** - Only left channel is used.\n- **Right** - Only right channel is used.\n- **Min** - The absolute minimum value is taken from stereo input.\n- **Max** - The absolute maximum value is taken from stereo input.\n\n**Low-Cut Filter** \nEnables a custom Low-Cut Filter for the selected band.\n\n**Low-Cut Frequency** \nSets the cut-off frequency of the custom Low-Cut filter. If it is disabled, the default Low-Cut filter assumes internally the Band Start frequency as the cut-off frequency.\n\n**Hight-Cut Filter** \nEnables a custom High-Cut Filter for the selected band.\n\n**Hight-Cut Frequency** \nSets the cut-off frequency of the custom High-Cut filter. If it is disabled, the default High-Cut filter assumes internally the Band End frequency as the cut-off frequency.\n\n**PreAmp** \nGain applied to the Sidechain signal of the selected band.\n\n**Reactivity** \nThe time that defines the number of samples used to process the Sidechain in RMS, Uniform and Low-Pass modes. Higher the value, more smooth the gating.\n\n**Lookahead** \nThe band signal to gate is delayed by this amount of time, so that the gating will be applied earlier than it would be otherwise.\n\nEach band can have different Lookahead values. To avoid phase distortions in the mixing stage, all the bands are automatically delayed for an individually calculated period of time.\n\n## References\n\n- [LSP Sidechain Multiband Gate Stereo](https://lsp-plug.in/?page=manuals§ion=sc_mb_gate_stereo)\n- [Wikipedia Noise Gate](https://en.wikipedia.org/wiki/Noise_gate)\n","src/contents/docs/plugins/pitch.md":"# Pitch\n\nPitch shifting is a sound recording technique in which the original Pitch of a sound is raised or lowered. Easy Effects uses the pitch shifter from SoundTouch.\n\n**Mode** \nControls the method used for pitch shifting.\n\n- **High Speed** - Uses a method with a CPU cost that is relatively moderate and predictable.\n- **High Quality** - Uses the highest quality method for pitch shifting. This CPU cost is approximately proportional to the required frequency shift.\n- **High Consistency** - Uses the method that gives greatest consistency when used to create small variations in pitch around the 1.0-ratio level. Unlike the previous two options, this avoids discontinuities when moving across the 1.0 pitch scale. It also consumes more CPU than the others in the case where the pitch scale is exactly 1.0.\n\n**Formant** \nControls the handling of formant shape (spectral envelope) when pitch-shifting.\n\n- **Shifted** - Applies no special formant processing. The spectral envelope will be pitch shifted as normal.\n- **Preserved** - Preserves the spectral envelope of the unshifted signal. This permits shifting the note frequency without so substantially affecting the perceived pitch profile of the voice or instrument.\n\n**Transients** \nControls the component frequency phase-reset mechanism that may be used at transient points to provide clarity and realism to percussion and other significant transient sounds.\n\n- **Crisp** - Resets component phases at the peak of each transient (the start of a significant note or percussive event). This usually results in a clear-sounding output, but it is not always consistent, and may cause interruptions in stable sounds present at the same time as transient events.\n- **Mixed** - Resets component phases at the peak of each transient, outside a frequency range typical of musical fundamental frequencies. The results may be more regular for mixed stable and percussive notes than Crisp option, but with a \"phasier\" sound. The balance may sound very good for certain types of music and fairly bad for others.\n- **Smooth** - Does not reset component phases at any point. The results will be smoother and more regular but may be less clear than the other transient options.\n\n**Detector** \nControls the type of transient detector used.\n\n- **Compound** - Uses a general purpose transient detector which is likely to be good for most situations.\n- **Percussive** - Detects percussive transients.\n- **Soft** - Uses an onset detector with less of a bias toward percussive transients. This may give better results with certain material (e.g. relatively monophonic piano music).\n\n**Phase** \nControls the adjustment of component frequency phases from one analysis window to the next during non-transient segments.\n\n- **Laminar** - Adjusts phases when stretching in such a way as to try to retain the continuity of phase relationships between adjacent frequency bins whose phases are behaving in similar ways. This should give good results in most situations.\n- **Independent** - Adjusts the phase in each frequency bin independently from its neighbours. This usually results in a slightly softer, phasier sound.\n\n**Cents** \nNumber of cents the Pitch will be increased or decreased.\n\n**Semitones** \nNumber of semitones the Pitch will be increased or decreased.\n\n**Octaves** \nNumber of octaves the Pitch will be increased or decreased.\n\n## References\n\n- [Wikipedia Pitch Shift](https://en.wikipedia.org/wiki/Pitch_shift)\n- [SoundTouch Audio Time Stretcher Library - Attractive Features](https://www.surina.net/soundtouch/)\n","src/contents/docs/plugins/reverb.md":"# Reverberation\n\nReverberation is the phenomenon of persistence of sound after the source has been stopped as a result of multiple reflections of the waves over objects within a closed surface. These reflections build up with each other and decay gradually as they are absorbed by the surfaces of objects in the space enclosed.\n\nThe Reverberation is different than Echo because the Echo is a reflected sound wave with sufficient magnitude and delay to be detectable as a signal distinct from the source one. To simulate the Reverberation effect, Easy Effects uses the Reverb plugin developed by Calf Studio Gear.\n\n**High Frequency Damping** \nCutoff frequency of the reflections. It causes higher frequencies to decay faster.\n\n**Room Size** \nSize of the space where simulated Reverberation occurs inside. It determines the time between reflections.\n\n**Diffusion** \nDegree of uniformity. Higher values lead to less uniform Reverberation.\n\n**Pre Delay** \nAdditional delay. It corresponds to a distance between sound source and the nearest wall.\n\n**Decay Time** \nThe time it takes for Reverberation to fade out.\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal (Reverberation) mixed in the output.\n\n**Bass Cut** \nRemoves low frequencies from the Reverberation.\n\n**Treble Cut** \nRemoves high frequencies from the Reverberation.\n\n## References\n\n- [Wikipedia Reverberation](https://en.wikipedia.org/wiki/Reverberation)\n- [Calf Reverb](https://calf-studio-gear.org/doc/Reverb.html)\n","src/contents/docs/plugins/rnnoise.md":"# Noise Reduction\n\nThe Noise Reduction is a process aimed to attenuate the disturbing noise from a signal.\n\nEasy Effects Noise Reduction is made on the RNNoise library which is based based on recurrent neural network, a class of artificial neural networks where connections between nodes form a directed graph along a temporal sequence. This allows it to exhibit temporal dynamic behavior.\n\nStandard RNNoise Model is used and custom models can be imported to perform different types of noise reduction.\n\n## References\n\n- [Wikipedia Noise Reduction](https://en.wikipedia.org/wiki/Noise_reduction)\n- [Wikipedia Recurrent Neural Network](https://en.wikipedia.org/wiki/Recurrent_neural_network)\n- [Jean-Marc Valin - RNNoise: Learning Noise Suppression](https://jmvalin.ca/demo/rnnoise/)\n","src/contents/docs/plugins/spectrum.md":"# Spectrum\n\n**Enabled** \nShow or hide the Spectrum.\n\n**Shape** \nChoose the style of the animation.\n\n**Points** \nNumber of points/bars on the x axis.\n\n**Height** \nChanges the height of the Spectrum.\n\n**Line Width** \nThe width of a single bar.\n\n**Fill** \nDraw filled bars.\n\n**Show Bars Border** \nDraw bars with borders.\n\n**Rounded Corners** \nDraw bars with rounded corners.\n\n**Color** \nChoose the color of the Spectrum components.\n\n**Minimum Frequency Range** \nLower end frequency of the Spectrum.\n\n**Maximum Frequency Range** \nUpper end frequency of the Spectrum.\n\n[Settings Menu](../user_interface/settingsmenu.md)","src/contents/docs/plugins/speex.md":"# Speech Processor\n\nThis plugin allows EasyEffects to use the Speex preprocessor to attenuate disturbing background noises from a signal.\n\nCompared to Noise Reduction which uses RNNoise to suppress noises, Speech Processor has the benefit of using less computational resources, at the cost of sacrificing noise suppression quality.\n\nFor more information on noise suppression in general, refer to the manual page on Noise Reduction.\n\n## References\n\n- [The Speex Project](https://www.speex.org/)\n","src/contents/docs/plugins/stereotools.md":"# Stereo Tools\n\nEasy Effects uses the Stereo Tools developed by Calf Studio Gear which provides some handy utilities to manage Stereo streams handling Left and Right channels in conjunction with Mid and Side signals.\n\nMid/Side is a recording technique which registers Mid and Side signals rather then plain Left and Right channels. Its basic setup makes use of one cardioid microphone (Mid) and one bidirectional (figure-eight) microphone (Sides). In mastering stage Mid and Side can be used to shape the balance of the Stereo image having more control over the mix to get a wider, deeper, and more focused track.\n\nHaving a Stereo stream, Mid is obtained summing both channels `(L+R)` while Side is the result of the difference between Left and Right `(L-R)`. Mid and Side signals can be decoded back to Left `(M+S)/2` and Right `(M-S)/2`.\n\n## Input\n\n**Balance** \nSets the balance between both channels.\n\n**Softclip (button)** \nMakes a kind of analog distortion instead of harsh digital 0 dB clipping.\n\n**Softclip (value)** \nLevel of Softclip.\n\n## Stereo Matrix\n\n**Mode** \nAllows to choose between different channel configurations such as normal Stereo Mode, Mid-Side or Mono simulation.\n\n**Mute L/R** \nMutes the Left/Right channel.\n\n**Invert Phase L/R** \nChanges the Phase of the Left/Right channel.\n\n**Side Level** \nThe level of the Side signal.\n\n**Side Balance** \nThe balance of the Side signal.\n\n**Middle Level** \nThe level of the Middle signal.\n\n**Middle Panorama** \nThe position in the Panorama of the Middle signal.\n\n## Output\n\n**Balance** \nSets the balance between both channels.\n\n**Delay L/R** \nDelays the Left or the Right channel. Negative values delay the Left channel and positive values the Right channel.\n\n**Stereo Base** \nSets the Stereo Base of the content seamless between Mono and inverted channels.\n\n**Stereo Phase** \nSet the Stereo Phase of the content.\n\n## References\n\n- [Wikipedia Stereophonic Sound](https://en.wikipedia.org/wiki/Stereophonic_sound)\n- [Calf Stereo Tools](https://calf-studio-gear.org/doc/Stereo%20Tools.html)\n- [Unlock Your Sound - Stereo, Mono, Mid, Side, Panning, and Imaging Explained](https://unlockyoursound.com/stereo/)\n","src/contents/docs/plugins/voicesuppressor.md":"# Voice Suppressor\n\nVery simple plugin that removes voice from audio. It calculates the correlation of the lefft and right channels in\nthe frequency domain. Frequencies whose correlation and phase difference are beyond the configured thresholds are attenuated. \n\n**Start** \n\nFrequencies below this value are not attenuated. \n\n**End** \n\nFrequencies above this value value are not attenuated. \n\n**Correlation** \n\nFrequencies are attenuated only if the correlation between left and right channel is at least equal to this value. \n\n**Phase Difference** \n\nFrequencies are not attenuated when the magnitude of phase difference between left and right channel is above this value. \n\n**Minimum Kurtosis** \n\nFrequencies with local kurtosis below this value are attenuated. \n\n**Maximum Instantaneous Frequency** \n\nThis is related to the frequency of rotation of the complex vector that represents the correlation between the left and right channel at a given frequency.\nIn other words this measures how fast the phase difference between the left and right channels changes at a given frequency bin in the Fourier transform.\nMore attenuation is applied to the channels samples that have rotation frequency below this parameter value. \n\n**Inverted Mode** \nInstead of suppressing voice the plugin will try to suppress the background and keep the voice. \n"},"files":{"src/contents/docs/plugins/autogain.md":"# Auto Gain\n\nEasy Effects Autogain is based on the library libebur128 which implements the EBU R 128 standard for loudness normalization. It changes the audio volume to a perceived loudness target that can be customized by the user.\n\n**Target** \nLoudness level.\n\n**Silence** \nSilence threshold. While the momentary loudness is below this value the autogain won't make any changes to the current gain being applied to the signal level.\n\n**Maximum History** \nRange of time taken into account for the calculation of loudness level and output gain.\n\n**Reference** \nThe parameter used as reference to evaluate the output gain.\n\n- **Momentary** - Measures the loudness of the past 400 milliseconds.\n- **Short-Term** - Measures the loudness of the past 3 seconds.\n- **Integrated** - Ideally it indicates how loud the content is on average. It measures the loudness on a long range of time, depending on the value set as Maximum History.\n- **Geometric Mean** - Uses the geometric mean of all the above mentioned parameters, or two of them at user choosing.\n\n**Reset History** \nResets the Autogain history related to chosen Reference.\n\n**Monitor Parameters** \nAutogain values shown as stats.\n\n- **Relative** - Used to detect silence. Whenever the Momentary term is below a predetermined threshold, modifications to the output gain will be disabled.\n- **Loudness** - The difference between its value and the target loudness determines the output gain.\n- **Range** - Indicates how large is the dynamic range of the content played.\n- **Output Gain** - The input signal is adjusted by this correction gain to bring its loudness to the target value.\n\n## References\n\n- [Wikipedia EBU R 128](https://en.wikipedia.org/wiki/EBU_R_128)\n- [EBU - Loudness Normalisation and Permitted Maximum Level of Audio Signals](https://tech.ebu.ch/publications/r128/)\n","src/contents/docs/plugins/autotune.md":"# Autotune\n\nAutotune is a pitch correction effect that automatically adjusts the pitch of an audio signal to the nearest target note. It can be used to subtly correct vocal intonation or to create the characteristic \"auto-tune effect\" heard in modern music. Easy Effects uses the fat1 plugin from x42 LV2 Plugins.\n\n## Tuning Options\n\n**Mode** \nControls how pitch correction is applied.\n\n- **Auto** - Automatically detects the pitch and corrects it to the nearest enabled note in the scale.\n- **Manual** - Targets a fixed pitch set by the Offset parameter.\n\n**Tuning** \nThe reference frequency for the A4 note, in Hz. The standard tuning is 440 Hz. Adjust this if you are working with non-standard tuning (range: 400–480 Hz).\n\n**Correction** \nControls the strength of pitch correction. At 1.0, the pitch is fully corrected to the target note. Lower values allow more of the natural pitch variation through, producing a subtler effect.\n\n**Bias** \nControls the preference toward the current note versus the nearest target note. At 0.5 (center), no bias is applied. Higher values make the correction stick to the current note longer before switching. Lower values make it switch to the nearest target more readily.\n\n**Filter** \nControls the smoothness of the pitch detection. Lower values result in faster, more responsive detection but may introduce artifacts. Higher values produce smoother, more stable detection but with slower response.\n\n**Offset** \nShifts the target pitch by the specified number of semitones (range: -2.0 to 2.0). In Manual mode, this sets the fixed target pitch offset.\n\n**Fast Correction** \nWhen enabled, applies pitch correction with minimal latency. This produces a more immediate, aggressive correction effect but may introduce more artifacts.\n\n## Scale Selection\n\nThe Scale section allows you to select which notes the pitch correction will target.\n\n**Root** \nThe root note of the scale (C, D, E, F, G, A, or B).\n\n**Accidental** \nApplies a sharp or flat modifier to the root note.\n\n**Scale** \nSelects either Major or Minor scale pattern. Choosing a scale automatically enables the correct notes for that key.\n\n**Note Toggles** \nIndividual toggles for each of the 12 chromatic notes (C through B). Selecting a scale sets these automatically, but you can also configure them manually for custom scales or chromatic correction (all notes enabled).\n\n## Pitch Error Meter\n\nDisplays the current pitch deviation from the target note. A value of 0 means the pitch is exactly on target. Negative values indicate the pitch is flat, positive values indicate it is sharp.\n\n## References\n\n- [x42 fat1 Auto Tune](https://x42-plugins.com/x42/x42-autotune)\n- [Wikipedia Auto-Tune](https://en.wikipedia.org/wiki/Auto-Tune)\n","src/contents/docs/plugins/bassenhancer.md":"# Bass Enhancer\n\nA Bass Enhancer is used to produce very low sound that is not present in the original signal. This is done by creating harmonic distortions of the signal which are restricted in range and added to the original signal. It raises the lower end of an audio signal without simply raising the lower frequencies like an [Equalizer](equalizer.md) would do to create a more \"fat\" or \"boomy\" sound.\n\n**Amount** \nAmount of harmonics added to the original signal.\n\n**Harmonics** \nAmount of newly created harmonics.\n\n**Scope** \nThe frequency above which harmonics are produced.\n\n**Floor (button)** \nConstrain the enhancement on the lower end.\n\n**Floor (value)** \nThe frequency below which no harmonics are produced.\n\n**Blend Harmonics** \nThe \"colour\" (or octave) of the harmonics.\n\n**Listen** \nMute the original signal and listen to the harmonics exclusively.\n\n## References\n\n- [Calf Bass Enhancer](https://calf-studio-gear.org/doc/Bass%20Enhancer.html)\n- [Wikipedia Missing Fundamental](https://en.wikipedia.org/wiki/Missing_fundamental)\n","src/contents/docs/plugins/bassloudness.md":"# Bass Loudness\n\nThe ear is less sensitive to low frequencies when listening at low volume. This plugin developed by MDA allows the bass level to be adjusted measuring the \"equal-loudness contour\".\n\n**Loudness** \nSource level relative to listening level (based on a 100 dB SPL maximum level).\n\n**Output** \nChange output level.\n\n**Link** \nAutomatically adjusts Output to maintain a consistent tonal balance at all levels.\n\n## References\n\n- [Wikipedia Equal-Loudness Contour](https://en.wikipedia.org/wiki/Equal-loudness_contour)\n- [MDA Loudness](http://mda.smartelectronix.com/vst/help/loudness.htm)\n","src/contents/docs/plugins/compressor.md":"# Compressor\n\nA Compressor is used to reduce the dynamic range or, in other words, the difference in level between the quietest and the loudest parts of an audio signal. It achieves this purpose altering the gain when the signal overtakes a predetermined Threshold. Easy Effects uses the Stereo Sidechain Compressor from Linux Studio Plugins.\n\n## Compressor Options\n\n**Attack Time** \nThe length of time it takes to apply roughly two-thirds of the targeted amount of compression ratio to the uncompressed signal.\n\n**Release Time** \nThe length of time it takes to restore roughly two-thirds of the reduced gain (in Downward mode) or increased gain (in Upward/Boosting mode) to the compressed signal.\n\n**Attack Threshold** \nThe target level around which the compression is applied (the range depends by the Knee).\n\n**Release Threshold** \nSets up the Threshold of the Release Time, calculated by summing the Release Threshold to the Attack Threshold.\n\nIf the Sidechain level is above that Threshold, the compressor uses the Release Time for the releasing stage. Otherwise the Attack Time is used in place of Release Time.\n\nFor example, with -10 dB Attack Threshold and -60 dB Release Threshold, if the Sidechain is above `-10 + (-60) = -70 dB`, the Release Time is used for the gain restoration. If the Sidechain is below -70 dB, the Attack Time is used for the gain restoration.\n\n**Ratio** \nThe amount of attenuation (in Downward mode) or amplification (in Upward/Boosting mode) that will be applied to the signal.\n\nFor example, when the Ratio is 2 in Downward mode and the Sidechain rises above the Threshold by 10 dB, the signal would be ideally reduced by 5 dB (10/2 dB). In practice this behavior mostly depends on how the Compressor is designed and configured.\n\n**Knee** \nThe range over which the Compressor switches from no compression to almost the full ratio compression (the Threshold usually sits at the center of this transition zone).\n\n**Makeup** \nThe gain to apply after the compression stage.\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal mixed in the output.\n\n**Mode** \n- **Downward** - It's aimed to decrease the gain of the signal above the Threshold.\n- **Upward** - It's aimed to increase the gain of the signal below the Threshold.\n- **Boosting** - It's aimed to increase the gain of the signal below the Threshold by a specific amount.\n\n**Boost Threshold** \nThe Threshold below which a constant amplification will be applied to the input signal in Upward Mode (it prevents from applying infinite amplification to very quiet signals).\n\n**Boost Amount** \nMaximum gain amplification to apply in Boosting Mode.\n\n## Sidechain\n\n**Listen** \nAllows to listen the processed Sidechain signal.\n\n**Input Type** \nDetermines which signal is the Sidechain or, in other words, the signal that controls the compression stage.\n\n- **Feed-forward** - The Sidechain is the Compressor input signal (taken after applying the plugin input gain). More aggressive compression.\n- **Feed-back** - The Sidechain is the Compressor output signal (taken before applying the Makeup and the plugin output gain). Vintage-style compression.\n- **External** - The Sidechain is an external source took by a specific input device (typically a microphone).\n\n**Input Device** \nSelect the device for the External Sidechain.\n\n**Mode** \nDetermines how the Sidechain is evaluated for the compression stage.\n\n- **Peak** - The Compressor reacts according to the peaks.\n- **RMS** - The Compressor reacts according to the average loudness measured by the root mean square.\n- **Low-Pass** - The Compressor reacts according to the signal processed by a Low-Pass filter.\n- **Uniform** - The Compressor reacts according to the loudness measured by the average of the absolute amplitude.\n\n**Source** \nDetermines which part of the Sidechain is taken into account for the compression stage.\n\n- **Middle** - The sum of left and right channels.\n- **Side** - The difference between left and right channels.\n- **Left** - Only left channel is used.\n- **Right** - Only right channel is used.\n- **Min** - The absolute minimum value is taken from stereo input.\n- **Max** - The absolute maximum value is taken from stereo input.\n\n**PreAmplification** \nGain applied to the Sidechain signal.\n\n**Reactivity** \nThe time that defines the number of samples used to process the Sidechain in RMS, Uniform and Low-Pass modes. Higher the value, more smooth the compression.\n\n**Lookahead** \nThe signal to compress is delayed by this amount of time, so that the compression will be applied earlier than it would be otherwise. The corresponding delay is reproduced on the output signal.\n\n## Sidechain Filters\n\n**High-Pass Filter Mode** \nSets the type of the High-Pass filter applied to Sidechain signal.\n\n**High-Pass Frequency** \nSets the cut-off frequency of the High-Pass filter.\n\n**Low-Pass Filter Mode** \nSets the type of the Low-Pass filter applied to Sidechain signal.\n\n**Low-Pass Frequency** \nSets the cut-off frequency of the Low-Pass filter.\n\n## References\n\n- [Wikipedia Dynamic Range Compression](https://en.wikipedia.org/wiki/Dynamic_range_compression)\n- [LSP Sidechain Compressor Stereo](https://lsp-plug.in/?page=manuals§ion=sc_compressor_stereo)\n- [Black Ghost Audio - The Ultimate Guide to Compression](https://www.blackghostaudio.com/blog/the-ultimate-guide-to-compression)\n- [Attack Magazine - Demolishing The Myths of Compression](https://www.attackmagazine.com/features/columns/gregory-scott-demolishing-the-myths-of-compression/)\n","src/contents/docs/plugins/convolver.md":"# Convolver\n\nThe Convolver creates a simulation of an audio environment using a pre-recorded audio sample of the impulse response of the space being modeled. This feature is based on the \"convolution\": a process through which the sonic characteristics of one signal are used to alter the character of another.\n\nEasy Effects Convolver offers the opportunity to apply multiple impulse responses by combining them in one file.\n\n**Impulses** \nManage impulse response files. This menu allows to load a specific impulse or add/remove files from the Easy Effects configuration directory.\n\n**Combine** \nSelect two impulse responses to combine into a single file to be saved under the Easy Effects configuration directory.\n\n**Stereo Width** \nModify the impulse response stereo image width.\n\n**Spectrum** \nVisualize the frequency spectrum of the selected channel.\n\n## References\n\n- [Wikipedia Reverb Effect](https://en.wikipedia.org/wiki/Reverb_effect)\n- [Indiana University - Convolution: A Form of Cross-Synthesis](https://cmtext.indiana.edu/synthesis/chapter4_convolution.php)\n- [Designing Sound - Recording Impulse Responses](https://designingsound.org/2012/12/29/recording-impulse-responses/)\n","src/contents/docs/plugins/crossfeed.md":"# Crossfeed\n\nEasy Effects uses the Crossfeed by bs2b. This plugin is used to improve headphone listening of stereo audio records. It does so by mixing the left and right channel in a way that simulates a stereo speaker setup while using headphones.\n\n**Cutoff** \nLow-Pass filter cutoff frequency.\n\n**Feed** \nAmount of signal from a channel that is sent to the other.\n\n**Presets** \n- **Default** - Closest to virtual speaker placement (30°, 3 meters).\n- **Cmoy** - Close to Chu Moy's Crossfeed (popular).\n- **Jmeier** - Close to Jan Meier's CORDA amplifiers (little change).\n\n## References\n\n- [Wikipedia Crossfeed](https://en.wikipedia.org/wiki/Crossfeed)\n- [Bauer Stereophonic-to-Binaural DSP.](http://bs2b.sourceforge.net/)\n","src/contents/docs/plugins/crosstalkcanceller.md":"# Crosstalk Canceller\n\nWhen audio is played by a stereo speaker audio from both channels arrive to the listener ears. As the superposition\nof each channel audio will happen with a delay that depends on the distance between the listener ear and the speakers\nundesirable effects can happen. That superposition is called crosstalk in this context and can be attenuated by this\nplugin.","src/contents/docs/plugins/crusher.md":"# Crusher\n\nA bitcrusher reduces the resolution or bandwidth of digital audio data. Audio reduced in bit depth sounds more harsh and \"digital\". \nIn the bitcrusher from Calf Studio Gear used in EasyEffects reduction can be done in a linear or logarithmic way.\nAccording to the plugin authors the logarithmic way results in a much smoother sound in low volume signals. \n\n**Mode** \n- **Linear** - Linear distance between bits.\n- **Logarithmic** - Logarithmic distances between bits. The result is a much more \"natural\" sounding crusher which doesn't gate low signals. \n\n**Bit reduction** \nControls the processed audio bit depth. \n\n**DC offset** \nThis offset causes different crushing of the lower and the upper half of the signal. \n\n**Anti-aliasing** \nControls the softness of the crushing sounds. \n\n**Mix** \nControls the mix of original and processed audio. When at 0% the output signal is equal to the original. When at 100% only the processed signal is in the output signal. \n\n**Reduction** \nControls the reduction of the sample rate (downsampling). \n\n**Low frequency oscillator active** \nEnable the low frequency oscillator. The oscillator will make the sample rate reduction change between a range determined by the range control.\n\n**Low frequency oscillator range** \nControls the amount of modulation applied to the sample rate reduction. \n\n**Low frequency oscillator rate** \nControls the frequency of the oscillator. \n\n## References\n\n- [Wikipedia Bit Crusher](https://en.wikipedia.org/wiki/Bitcrusher)\n- [Calf Crusher](https://calf-studio-gear.org)\n- [Calf Crusher Documentation](https://calf-studio-gear.org/doc/Crusher.html)\n","src/contents/docs/plugins/crystalizer.md":"# Crystalizer\n\nThe Crystalizer plugin can be used to add a little of dynamic range to songs that were overly compressed. The signal is split in multiple bands to which different intensities can be applied in order to alter the overall dynamic range.\n\n**Intensities**\nThe higher the value the higher is the difference in magnitude between the loudest and the quietest sounds of the selected band. Different intensities can be set for each frequency band.\n\n**Bypass**\nWhen active the audio signal passing through the selected band is not modified.\n\n**Mute**\nMutes the selected band.\n\n**Adaptive Intensity**\nWhen enabled the intensity value set by the band slider is continuously scaled as audio is processed. The scaling may\nincrease or decrease the intensity based on the geometric mean of the signal crest factor, kurtosis and spectral flux.\n\nWhen using static intensity values it may happen that the crystalizer enhances high frequency audio too much. What leads\nto audible artifacts. The adaptive scaling removes this is most of the cases.\n\n**Oversampling**\nResamples the signal to double its sampling rate value and back to the original value after the plugins effects are applied.\nIt improves the effects quality but it also increases CPU usage.\n\n**Oversampling Quality**\nControls the resampling quality. Higher values require more CPU power.\n\n**Fixed Quantum**\nForces the plugin to use a fixed audio buffer size to process audio. This helps to avoid noises when the sound server switches latency on the fly.\nBut it has the downside of introducing unneeded latency in some cases. And increased CPU usage in cases where higher latency could be used.\n\n**Transition Band**\nControls the width of the transition band of the bandpass filters used to split the signal in several bands. Higher values will use less CPU. But will result in more overlap between the bands.\n\n## References\n\n- [Wikipedia Dynamic Range](https://en.wikipedia.org/wiki/Dynamic_range)\n- [Crest factor](https://en.wikipedia.org/wiki/Crest_factor)\n- [Kurtosis](https://en.wikipedia.org/wiki/Kurtosis)\n- [Spectral flux](https://en.wikipedia.org/wiki/Spectral_flux)\n","src/contents/docs/plugins/deepfilternet.md":"# Deep Noise Remover\n\nAdvanced noise reduction is achieved using \"DeepFilterNet\" technology. This technology uses AI technology called deep learning to efficiently remove noise from audio signals, enabling higher quality noise suppression than conventional methods. This technology has several control parameters, which are explained below and how to set them.\n\n**Model** \nSelects the noise reduction model. Each one must be installed separately.\n\n- DeepFilterNet: requires [DeepFilterNet](https://github.com/Rikorose/DeepFilterNet).\n- DPDFNet: requires [HushMic](https://github.com/Fovty/HushMic). The Attenuation Limit and Process as Mono controls apply to this model.\n\n**Process as Mono (DPDFNet only)**\n\nAverages the left and right input channels before resampling, processes the result with one model instance, and sends the same processed signal to both output channels. This can significantly reduce CPU usage for mono microphones that supply the same signal on both channels. It does not change the selected model or its noise reduction quality.\n\nThis setting is off by default and is saved in presets. Enable it for mono speech input; leave it off when stereo separation matters. If only one input channel contains audio, averaging reduces its level by 6 dB. Opposite-polarity signals can cancel, so check the input channels before enabling it.\n\n**Attenuation Limit** \nThis parameter determines how much noise is attenuated. Higher values result in stronger noise reduction, but too high a value may also remove parts of the audio.\n\n- Recommended setting: 70dB to 80dB is a typical balanced value. This minimizes background noise while preserving speech intelligibility.\n\n**Minimum Processing Threshold** \nSignals below the Minimum Processing Threshold are targeted for denoising. Signals louder than this threshold are not subject to noise reduction and are passed through unchanged. This setting is intended to effectively remove small background noise.\n\n- Recommended setting: -15dB to -30dB is good. This should be fine-tuned according to the ambient noise. Currently, Easy Effects does not allow a value smaller than -15dB.\n\n**Maximum ERB Processing Threshold** \nThe human auditory system has the ability to separate and recognize different frequency components. Equivalent Rectangular Bandwidth (ERB) is a numerical expression and model of this filtering capability. ERB-based processing emphasizes the perceptual importance of speech by focusing on specific frequency bands. This improves the effectiveness of noise reduction and speech enhancement. This threshold sets the maximum level at which perceptually equivalent bandwidth (ERB)-based processing is applied. Signals above this threshold are not subject to noise reduction or speech enhancement processing. This prevents important speech components or loud portions of speech from being unnecessarily processed.\n\n- Recommended setting: Start in the range of 20dB to 30dB and adjust as needed.\n\n**Maximum DF Processing Threshold** \nThe Maximum DF Processing Threshold sets the maximum level of signal that DeepFilterNet will process. Signals exceeding this threshold are excluded from noise reduction processing. This prevents important or loud portions of speech from being unnecessarily processed.\n\n- Recommended setting: 20 dB as the default value. This effectively suppresses noise while maintaining voice quality.\n\n**Minimum Processing Buffer** \nDefines the size of the buffer used to process frames of audio data. A larger buffer can improve noise suppression but may increase latency.\n\n- Recommended setting: Set to a small value (0) if real-time processing is important, or to a number greater than 1 if noise suppression accuracy is a priority.\n\n**Post Filter Beta** \nControls the intensity of the post-processing filter applied after the initial noise suppression. This allows for more subtle refinement of the audio signal.\n\n- Recommended setting: 0.5dB to 2dB is generally recommended, but currently Easy Effects has a maximum value of 0.05dB, which is almost ineffective.\n","src/contents/docs/plugins/deesser.md":"# Deesser\n\nA Deesser is used to dynamically reduce high frequencies. The standard field of use of this plugin is the reduction of \"sssss\" and \"shhhh\" in vocal tracks. Easy Effects uses the Deesser developed by Calf Studio Gear.\n\n**Detection** \nSelect the detection of the Sidechain signal between Peak (stronger) and RMS (smoother).\n\n**Mode** \nSelect the operation mode between Wideband and Split. In Split mode not the full range signal will be affected by the gain reduction, but only frequencies above the split frequency will be manipulated in gain.\n\n**F1 Split** \nThe split frequency. All signals above this frequency will affect the gain reduction (and are affected in Split mode too).\n\n**F1 Gain** \nIt shifts the volume of the higher band. In Wideband mode it affects the Sidechain. In Split mode it also affects the processed high frequencies.\n\n**F2 Peak** \nCenter frequency of the bell filter. It allows a more precise selection of the Sidechain signal.\n\n**F2 Level** \nIncreases or decreases the level of the chosen F2 frequency.\n\n**F2 Peak Q** \nSet the quality of the bell filter. Higher values will affect a narrower frequency range. Lower values will affect a wider band.\n\n**Laxity** \nThe reaction of the Deesser. Higher values won't affect really short peaks.\n\n**Threshold** \nThe level above which the gain reduction is applied.\n\n**Ratio** \nThe amount of attenuation applied to the signal.\n\nFor example, a Ratio of 2 means that if the level rises 4 dB above the Threshold, it will be only 2 dB above after the reduction.\n\n**Makeup** \nThe gain to apply after the processing stage. In Split mode only the high band will be made up.\n\n## References\n\n- [Wikipedia De-essing](https://en.wikipedia.org/wiki/De-essing)\n- [Calf Deesser](https://calf-studio-gear.org/doc/Deesser.html)\n- [LedgerNote - De-esser: The Guide for Sibilant-Free Vocal Recordings](https://ledgernote.com/columns/mixing-mastering/de-esser/)\n","src/contents/docs/plugins/delay.md":"# Delay\n\nThis plugin allows the user to add a short Delay of time to each individual channel of the stereo stream. Easy Effects Delay uses the Delay Compensator Stereo by Linux Studio Plugins set in time mode.\n\n**Left Delay** \nLeft channel delay in milliseconds.\n\n**Right Delay** \nRight channel delay in milliseconds.\n\n**Left Dry Level** \nAmount of right unprocessed signal mixed in the output.\n\n**Right Dry Level** \nAmount of right unprocessed signal mixed in the output.\n\n**Left Wet Level** \nAmount of left processed signal mixed in the output.\n\n**Right Wet Level** \nAmount of right processed signal mixed in the output.\n\n## References\n\n- [Wikipedia Delay (audio effect)](https://en.wikipedia.org/wiki/Delay_(audio_effect))\n- [LSP Delay Compensator Stereo](http://lsp-plug.in/?page=manuals§ion=comp_delay_x2_stereo)\n","src/contents/docs/plugins/echocanceller.md":"# Echo Canceller\n\nThe Echo is a reflected sound wave with sufficient magnitude and delay to be detectable as a signal distinct from the source one. An Echo Canceller is used to improve voice quality by preventing Echo from being created or removing it after it has been added to the source signal. Easy Effects uses the Echo Canceller from SpeexDSP library.\n\n**Frame Size** \nThe amount of time in milliseconds to process at once. It is recommended to use a frame size in the order of 20 ms.\n\n**Filter Length** \nThe amount of time of the Echo cancelling filter to use (also known as tail length). The recommended tail length is approximately the third of the room reverberation time. For example, in a small room, reverberation time is in the order of 300 ms, so a tail length of 100 ms is a good choice.\n\n## References\n\n- [Wikipedia Echo Suppression and Cancellation](https://en.wikipedia.org/wiki/Echo_suppression_and_cancellation)\n- [Speex Acoustic Echo Canceller](https://www.speex.org/docs/manual/speex-manual/node4.html#SECTION00450000000000000000)\n","src/contents/docs/plugins/equalizer.md":"# Equalizer\n\nThe Equalization in sound recording and reproduction is the process of adjusting the volume of different frequency bands within an audio signal. Easy Effects uses the Parametric Equalizer from Linux Studio Plugins. The user can choose from 1 to 32 bands. Width and center frequency of each band can be customized as needed.\n\n## Global Options\n\n**Bands** \nThe number of bands.\n\n**Mode** \n- **IIR** - Infinite Impulse Response filters, nonlinear minimal phase. In most cases does not add noticeable latency to output signal.\n- **FIR** - Finite Impulse Response filters with linear phase, finite approximation of Equalizer's impulse response. Adds noticeable latency to output signal.\n- **FFT** - Fast Fourier Transform approximation of the frequency chart, linear phase. Adds noticeable latency to output signal.\n\n**Balance** \nBalance between left and right output channels.\n\n**Pitch Left** \nThe frequency shift for all filters of the left channel, in semitones.\n\n**Pitch Right** \nThe frequency shift for all filters of the right channel, in semitones.\n\n**Split Channels** \nWhen enabled it is possible to apply different configurations to left and right channels.\n\n**Flat Response** \nThis function sets each band gain to 0.\n\n**Calculate Frequencies** \nThis function calculates the center frequency and the width of each band using the current number of bands. Useful when the user wants fewer than 32 bands but has no idea about which frequencies should be chosen.\n\n## Band Options\n\n**Type** \n- **Off** - The filter is not working (turned off).\n- **Bell** - Bell filter with smooth peak/recess.\n- **High Pass** - High Pass filter with rejection of low frequencies.\n- **High Shelf** - Shelving filter with adjustment of high frequency range.\n- **Low Pass** - Low Pass filter with rejection of high frequencies.\n- **Low Shelf** - Shelving filter with adjustment of low frequency range.\n- **Notch** - Notch filter with full rejection of selected frequency.\n- **Resonance** - Resonance filter with sharp peak/recess.\n- **All Pass** - All Pass filter.\n\n**Mode** \n- **RLC** - Very smooth filters based on similar cascades of RLC contours. Bilinear Z-transform (BT) or Matched Z-transform (MT) is used for pole/zero mapping.\n- **BWC** - Butterworth-Chebyshev-type-1 based filters. Does not affect Resonance and Notch filters. Bilinear Z-transform (BT) or Matched Z-transform (MT) is used for pole/zero mapping.\n- **LRX** - Linkwitz-Riley based filters. Does not affect Resonance and Notch filters. Bilinear Z-transform (BT) or Matched Z-transform (MT) is used for pole/zero mapping.\n- **APO** - Digital biquad filters derived from canonic analog biquad prototypes digitalized through Bilinear transform. These are [textbook filters](https://shepazu.github.io/Audio-EQ-Cookbook/audio-eq-cookbook.html) which are implemented as in the [Equalizer APO](https://equalizerapo.com/) software. Direct design (DR) is used to serve the digital filter coefficients directly in the digital domain, without performing transforms.\n\n**Slope** \nThe slope of the filter characteristics.\n\n**Solo** \nMakes the selected band the only one active.\n\n**Mute** \nMutes the selected band.\n\n**Frequency** \nCenter frequency of the selected band.\n\n**Width** \nBandwidth calculated as `width = frequency / quality`.\n\n**Quality** \nThe quality factor of the filter used.\n\n## References\n\n- [Wikipedia Equalization (audio)](https://en.wikipedia.org/wiki/Equalization_(audio))\n- [LSP Parametric Equalizer x32 LeftRight](http://lsp-plug.in/?page=manuals§ion=para_equalizer_x32_lr)\n- [Wikipedia Q Factor](https://en.wikipedia.org/wiki/Q_factor)\n- [How to EQ - Q Factor and Bandwidth in EQ: What They Mean](https://howtoeq.wordpress.com/2010/10/07/q-factor-and-bandwidth-in-eq-what-it-all-means/)\n","src/contents/docs/plugins/exciter.md":"# Exciter\n\nAn Exciter is used to produce high sound that is not present in the original signal. This is done by creating harmonic distortions of the signal which are restricted in range and added to the original signal. It raises the upper end of an audio signal without simply raising the higher frequencies like an [Equalizer](equalizer.md) would do to create a more \"crisp\" or \"brilliant\" sound.\n\n**Amount** \nAmount of harmonics added to the original signal.\n\n**Harmonics** \nAmount of newly created harmonics.\n\n**Scope** \nThe frequency above which harmonics are produced.\n\n**Ceiling (button)** \nConstrain the excitement on the upper end.\n\n**Ceiling (value)** \nThe frequency above which no harmonics are produced.\n\n**Blend Harmonics** \nThe \"colour\" (or octave) of the harmonics.\n\n**Listen** \nMute the original signal and listen to the harmonics exclusively.\n\n## References\n\n- [Calf Exciter](https://calf-studio-gear.org/doc/Exciter.html)\n- [Wikipedia Exciter (effect)](https://en.wikipedia.org/wiki/Exciter_(effect))\n","src/contents/docs/plugins/expander.md":"# Expander\n\nAn expander is an effect that works inversely to a compressor, further reducing the volume of quiet sounds below a certain level. This allows for natural suppression of background noise and unwanted whispers. For example, it is effective when air conditioning or other environmental sounds intrude into quiet scenes.\n\nThis effect is similar to the gate, which is explained in another chapter, in that it blocks quiet sounds. However, while a gate completely cuts sound, an expander gradually reduces sound, resulting in a more natural and smooth decay characteristic. Its main uses are listed below:\n\n- Reduce the prominence of quiet environmental noises (air conditioning, PC fans, clicks, etc.) during periods of silence\n- Suppress noise in quiet sections while maintaining the clarity of narration and dialogue\n- Emphasize the contours (punch) of sounds to create a more dynamic impression\n\n## Expander Options\n\nThe following is an explanation of each parameter when the mode is set to \"**Downward**.\"\n\n**Attack Time**: This determines the time it takes for the expander to begin suppressing the sound after the sound volume drops below the threshold. Shorter values will reduce the volume immediately, quickly suppressing noise and other imperfections. Longer values will cause the processing to begin later, preserving the natural reverberation and nuances of soft sounds.\n\n**Release Time**: This determines the time it takes for the expander to stop suppressing the volume and return to its original volume after the sound exceeds the threshold again and is determined to no longer require processing.\n\nLonger values will cause the sound to return more slowly, making it sound more natural.\n\nShorter values will return the sound to its original volume quickly, making it sound clearer and tighter, but may sound unnatural in some cases.\n\n**Attack Threshold**: The effect is only applied to sounds quieter than this value.\n\n**Release Threshold**: This can be the same as the Attack Threshold, but setting it about -5dB lower than the Attack Threshold will enable the Hysteresis function, suppressing fluttering around the threshold and achieving more stable processing.\n\n**Ratio**: This is an indicator of how much quieter (= how much) sounds below the threshold will be reduced. Set the value using the following as a guide:\n\n- A setting of 1 passes the signal as is (same as Expander OFF)\n- For light noise suppression (natural processing), use 1.5-3\n- For clarity of dialogue and smoothing out quiet sections, use 4-8\n- For complete noise reduction (gate-like behavior), use 20-100\n\n*Please note that setting the ratio too high can result in a choppy, unnatural sound.\n\n**Knee**: This parameter adjusts how smoothly processing begins near the threshold. The name comes from the fact that the curve on the graph resembles a knee. See below for the relationship between the setting value and the curve.\n\nKnee Value Volume Curve Characteristics Processing Sensation\n\n0dB Abrupt bend (hard knee) The effect is applied suddenly (like an ON/OFF switch)\n\n-6dB Gradual bend Processing begins naturally\n\n-12dB and above Very smooth Processing is barely noticeable\n\n**Makeup**: The gain applied after gating.\n\n**Dry Level**: The amount of unprocessed signal mixed with the output.\n\n**Wet Level**: The amount of processed signal mixed with the output.\n\nFor information on how to set up sidechaining and sidechain filtering, see the chapter of Gate.\n\n*When \"**Upward**\" is selected in the mode setting, the expander will function to boost (increase the volume of) quieter sounds.\n\nThis can be used in situations where dialogue or narration is too quiet to be heard, and is also effective when you want to bring out the subtle nuances of a musical performance (such as pianissimo).\n\nRather than \"lowering louder sounds\" like a compressor, it is used to increase the average volume by \"boosting quieter sounds,\" or to increase sound pressure while maintaining the punch and three-dimensionality of the original sound.","src/contents/docs/plugins/filter.md":"# Filter\n\nA Filter is used to amplify (boost), pass or attenuate (cut) defined parts of a frequency spectrum. Easy Effects uses the Filter from Linux Studio Plugins.\n\n**Type** \n- **Low-pass** - Low-pass filter with rejection of high frequencies. \n- **High-pass** - High-pass filter with rejection of low frequencies. \n- **Low-shelf** - Shelving filter with adjustment of low frequencies. \n- **High-shelf** - Shelving filter with adjustment of high frequency range. \n- **Bell** - Bell filter with smooth peak/recess. \n- **Bandpass** - Bandpass filter. \n- **Notch** - Notch filter with full rejection of selected frequency. \n- **Resonance** - Resonance filter with sharp peak/recess. \n- **Ladder-pass** - The filter that makes some ladder-passing in the spectrum domain. \n- **Ladder-rej** - The filter that makes some ladder-rejection in the spectrum domain. \n- **Allpass** - All-pass filter which only affects the phase of the audio signal at the specified frequency. \n\n**Filter Mode** \n- **RLC** - Very smooth filters based on similar cascades of RLC contours. \n- **BWC** - Butterworth-Chebyshev-type-1 based filters. Does not affect Resonance and Notch filters. \n- **LRX** - Linkwitz-Riley based filters. Does not affect Resonance and Notch filters. \n- **APO** - Digital biquad filters derived from canonic analog biquad prototypes digitalized through Bilinear transform. These are textbook filters which are implemented as in the EqualizerAPO software. \n- **BT** - Bilinear Z-transform is used for pole/zero mapping. \n- **MT** - Matched Z-transform is used for pole/zero mapping. \n- **DR** - Direct design is used to serve the digital filter coefficients directly in the digital domain, without performing transforms. \n\n**Equalizer Mode** \n- **IIR** - Infinite Impulse Response filters, nonlinear minimal phase. In most cases does not add noticeable latency to output signal. \n- **FIR** - Finite Impulse Response filters with linear phase, finite approximation of equalizer's impulse response. Adds noticeable latency to output signal. \n- **FFT** - Fast Fourier Transform approximation of the frequency chart, linear phase. Adds noticeable latency to output signal. \n- **SPM** - Spectral Processor Mode of equalizer, equalizer transforms the magnitude of signal spectrum instead of applying impulse response to the signal.\n\n**Slope** \nThe slope of the filter characteristics. \n\n**Frequency** \nThe cutoff/resonance frequency of the filter or the middle frequency of the band. \n\n**Width** \nThe width of the bandpass/ladder filters in octaves. \n\n**Gain** \nThe gain of the filter. It is disabled for lo-pass/hi-pass/notch filters. \n\n**Quality** \nThe quality factor of the filter. \n\n**Balance** \nThe balance between left and right output channels. \n\n## References\n\n- [Wikipedia Audio Filter](https://en.wikipedia.org/wiki/Audio_filter)\n- [Linux Studio Plugins Filter](https://lsp-plug.in/?page=manuals§ion=filter_stereo)\n","src/contents/docs/plugins/gate.md":"# Gate\n\nThe Gate attenuates signals that register below a Threshold. This kind of signal processing is used to reduce disturbing noise between useful signals. Easy Effects uses the Stereo Sidechain Gate from Linux Studio Plugins.\n\n## Gate Workflow\n\nThe Gate begins to open when the Sidechain level becomes above the Attack Zone Start level.\n\nThe Gate fully opens when the Sidechain level becomes above the Attack Threshold level.\n\nThe Gate begins to close when the Sidechain level becomes below the Release Zone Start level.\n\nThe Gate fully closes when the Sidechain level becomes below the Release Threshold level.\n\n## Gate Options\n\n**Attack Time** \nThe length of time it takes to restore roughly two-thirds of the gain reduction.\n\n**Release Time** \nThe length of time it takes to apply roughly two-thirds of the gain reduction.\n\n**Curve Threshold** \nThe Gate fully opens upon the Sidechain level becoming above Curve Threshold (displayed as Attack Threshold level).\n\nIf Hysteresis is not enabled, the Gate begins to close upon the Sidechain level becoming below Curve Threshold (displayed as Release Zone Start level).\n\n**Curve Zone Size** \nThe Gate begins to open upon the Sidechain level becoming above the Curve Threshold + Curve Zone (displayed as Attack Zone Start level).\n\nIf Hysteresis is not enabled, the Gate fully closes upon the Sidechain level becoming below the Curve Threshold + Curve Zone (displayed as Release Threshold level).\n\n**Hysteresis** \nWhen enabled, Curve Threshold and Curve Zone apply only to the opening Gate, and separate parameters can be configured for closing Gate.\n\n**Hysteresis Threshold** \nIf Hysteresis is enabled, the Gate begins to close upon the Sidechain level becoming below Curve Threshold + Hysteresis Threshold (displayed as Release Zone Start level).\n\n**Hysteresis Zone Size** \nIf Hysteresis is enabled, the Gate fully closes upon the Sidechain level becoming below the Curve Threshold + Hysteresis Threshold + Hysteresis Zone (displayed as Release Threshold level).\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal mixed in the output.\n\n**Reduction** \nIf the value is negative, it acts as the amount of gain reduction to apply to the input signal when the Gate is fully closed. If the value is positive, the Gate operates in \"Reverse Mode\": It reduces the amplitude when the signal is above the threshold.\n\n**Makeup** \nThe gain to apply after the gating stage.\n\n## Sidechain\n\n**Listen** \nAllows to listen the processed Sidechain signal.\n\n**Input Type** \nDetermines which signal is the Sidechain or, in other words, the signal that controls the gating stage.\n\n- **Internal** - The Sidechain is the Gate input signal (taken after applying the plugin input gain).\n- **External** - The Sidechain is an external source took by a specific input device (typically a microphone).\n\n**Input Device** \nSelect the device for the External Sidechain.\n\n**Mode** \nDetermines how the Sidechain is evaluated for the gating stage.\n\n- **Peak** - The Gate reacts according to the peaks.\n- **RMS** - The Gate reacts according to the average loudness measured by the root mean square.\n- **Low-Pass Filter** - The Gate reacts according to the signal processed by a recursive 1-pole Low-Pass filter.\n- **Simple Moving Average** - The Gate reacts according to the signal processed by the Simple Moving Average filter.\n\n**Source** \nDetermines which part of the Sidechain is taken into account for the gating stage.\n\n- **Middle** - The sum of left and right channels.\n- **Side** - The difference between left and right channels.\n- **Left** - Only left channel is used.\n- **Right** - Only right channel is used.\n- **Min** - The absolute minimum value is taken from stereo input.\n- **Max** - The absolute maximum value is taken from stereo input.\n\n**PreAmplification** \nGain applied to the Sidechain signal.\n\n**Reactivity** \nThe time that defines the number of samples used to process the Sidechain in RMS, Uniform and Low-Pass modes. Higher the value, more smooth the gating.\n\n**Lookahead** \nThe signal to gate is delayed by this amount of time, so that the gating will be applied earlier than it would be otherwise. The corresponding delay is reproduced on the output signal.\n\n## Sidechain Filters\n\n**High-Pass Filter Mode** \nSets the type of the High-Pass filter applied to Sidechain signal.\n\n**High-Pass Frequency** \nSets the cut-off frequency of the High-Pass filter.\n\n**Low-Pass Filter Mode** \nSets the type of the Low-Pass filter applied to Sidechain signal.\n\n**Low-Pass Frequency** \nSets the cut-off frequency of the Low-Pass filter.\n\n## References\n\n- [Wikipedia Dynamic Range Compression](https://en.wikipedia.org/wiki/Dynamic_range_compression)\n- [LSP Sidechain Gate Stereo](https://lsp-plug.in/?page=manuals§ion=sc_gate_stereo)\n- [Wikipedia Noise Gate](https://en.wikipedia.org/wiki/Noise_gate)\n","src/contents/docs/plugins/harmonizer.md":"# Harmonizer\n\nThe Harmonizer adds two pitch shifted voices to the input. Set the root note and chord, then choose an interval and level for each voice. The dry/wet control balances the original signal with the harmonized signal.\n\nEasy Effects uses the StereoHarm LV2 plugin from Rakarrack Plus. Install Rakarrack Plus to use this effect.\n\n## References\n\n- [Rakarrack Plus](https://github.com/Stazed/rakarrack-plus)\n- [Harmonizer documentation](https://github.com/Stazed/rakarrack-plus)\n","src/contents/docs/plugins/limiter.md":"# Limiter\n\nA Limiter is a special type of downward [Compressor](compressor.md) which does not allow the signal to overtake a predetermined Threshold. Ideally it has a very high compression ratio that takes the amplitude below a ceiling which stands as the maximum output level. For this reason it is usually named \"brick-wall limiter\".\n\nEasy Effects uses the Sidechain Stereo Limiter from Linux Studio Plugins. In most cases it works as a brick-wall limiter, but it offers also an additional feature that acts like a Compressor with extreme settings, so the output signal may exceed the specified Threshold.\n\n**Mode** \nSelect the operative mode of the peak cutting algorithm which searches the peaks above the Threshold and applies short gain reduction patches to the signal.\n\nThese patches can be selected in 3 forms: **Hermite**, **Exponential** and **Linear**. Each one has 4 different variants related to gain reduction of the samples around the peak: **Thin**, **Tail**, **Duck** and **Wide**.\n\nThe shape of each form and variant can be referred into the Linux Studio Plugin manual. See References section at the bottom of the present document.\n\n**Oversampling** \nWhen enabled, the sample rate is internally increased in order to improve peak detection and reduce aliasing (that causes distortion).\n\nThe modes have 2 main types: **Full**, which increases both the Sidechain and the Input signals, and **Half** which increases only the Sidechain. Each one of them specifies different **multipliers** and, between parentheses, the number of \"**lobes** in the kernel\".\n\nThe oversampled Input signal is downsampled to the original sample rate after processing.\n\n**Dither** \nIf specified, it enables Dithering for the selected bit depth. The process of Dithering adds a low-level noise to output signal in order to mask \"quantization distortion\", a form of artifact present in digital audio rendered at lower bit depth.\n\n**SC PreAmp** \nThe gain applied to the Sidechain before it is processed.\n\n**Lookahead** \nThe size of the buffer used to detect the peaks in advance. It adds the corresponding latency to the output signal.\n\n**Attack** \nThe length of time it takes to apply the needed gain reduction to keep the peak below the Threshold.\n\nIt affects the length of the gain reduction patch. It cannot be larger than the Lookahead (if specified larger, it's set internally as the max possible value).\n\n**Release** \nThe length of time it takes to restore the reduced gain around the limited peak.\n\nIt affects the length of the gain reduction patch. It cannot be twice larger than the Lookahead (if specified larger, it's set internally as the max possible value).\n\n**Threshold** \nThe target level above which the Limiter should reduce the peak of the signal. In some modes it represents the maximum output level.\n\n**Threshold Boost** \nIf enabled it applies an amount of gain to the limited signal equal to the absolute value of the Threshold. This causes the peak limited at the Threshold level to output at 0 dB (it has only effect when the Threshold is set below 0 dB).\n\n**Stereo Link** \nThe degree of the channel linking. At 0% both channels are limited independently while at 100% the loudest one triggers the same gain reduction on both.\n\n**External Sidechain** \nSwitch the Sidechain to an external source took by a specific input device (typically a microphone).\n\n## Auto Leveling\n\nThe Auto Leveling checkbutton introduces an additional feature named \"Automated Level Regulation\" (ALR) which acts like a Compressor with infinite ratio for the purpose of applying a smoothed gain reduction rather than a stronger peak cutter like in brick-wall mode. This configuration could get the output level to exceed the Threshold even if the signal is highly compressed.\n\n**Attack** \nManage how the raise of the input signal affects the smoothness of the ALR curve that controls the gain reduction level. Higher the value, more quickly the curve goes to it's maximum.\n\n**Release** \nManage how the fall of the input signal affects the smoothness of the curve that controls the gain reduction level. Higher the value, more quickly the curve goes to it's minimum.\n\n**Knee** \nManage the Threshold of the ALR gain curve and, in fact, adjust the balance between two gain reduction stages. Raising the value delegates more work to the peak-cutting algorithm. Lowering the value delegates more work to the ALR gain reduction algorithm.\n\n## References\n\n- [Wikipedia Limiter](https://en.wikipedia.org/wiki/Limiter)\n- [LSP Sidechain Limiter Stereo](https://lsp-plug.in/?page=manuals§ion=sc_limiter_stereo)\n","src/contents/docs/plugins/loudness.md":"# Loudness\n\nEasy Effects uses the Loudness Compensator from Linux Studio Plugins which applies the \"equal-loudness contour\" corrections to the input signal.\n\nAn equal-loudness contour is a measure of sound pressure level (SPL), over the frequency spectrum, for which a listener perceives a constant loudness when presented with pure steady tones. The unit of measurement for loudness levels is the phon and is arrived at by reference to equal-loudness contours.\n\nUsage of equal-loudness contours solves many mixing problems that every sound engineer meets while working on the track. The main problem is that human ear perceives different frequencies for different volume settings in a different way. In other words, applying changes to the mix on the low volume settings may cause unexpected sounding of the mix at the maximum loudness.\n\nThe Loudness Compensator performs frequency response computations and applies the computed frequency response to the input signal depending on the output volume settings. Additionally it can provide ear protection by applying hard-clipping to the output signal if it exceeds the allowed configurable level.\n\n**Standard** \nAllows to select different equal-loudness contour standards.\n\n- **Flat** - Applies flat frequency response to the whole spectrum. It's similar to just a gain knob but useful to perform a comparison to other modes.\n- **ISO 226:2003** - Recent equal-loudness contour standard published in 2003.\n- **Fletcher-Munson** - The first equal-loudness contour implementation by Harvey Fletcher and Wilden A. Munson published in 1933.\n- **Robinson-Dadson** - More accurate equal-loudness contour implementation by D.W. Robinson and R.S. Dadson published in 1956. It became the basis for the ISO 226:2003 standard.\n\n**FFT Size** \nAllows to select size of the Fast Fourier Transform frame used for the processing. The larger FFT frame is, the more precise the curve is approximated and the more latency is introduced.\n\n**Output Volume** \nThe output volume of the signal with applied equal loudness contour. It controls the loudness of the 1 kHz pure sine wave.\n\n**Clipping** \nAllows to enable and the hard clipping of the output signal.\n\n**Clipping Range** \nAllows to set the gap level for the hard clipping of the output signal.\n\n## References\n\n- [Wikipedia Equal-Loudness Contour](https://en.wikipedia.org/wiki/Equal-loudness_contour)\n- [LSP Loudness Compensator Stereo](https://lsp-plug.in/?page=manuals§ion=loud_comp_stereo)\n- [Lindos Electronics - Equal-Loudness Contours](http://www.lindos.co.uk/cgi-bin/FlexiData.cgi?SOURCE=Articles&VIEW=full&id=17)\n","src/contents/docs/plugins/maximizer.md":"# Maximizer\n\nThe Maximizer is a special type of [Limiter](limiter.md) that does not only prevent the signal to exceed a specified target level, but also adjusts the average Loudness of the audio track.\n\nEasy Effects uses the Maximizer developed by ZamAudio. It acts like an Amplifier that feeds a brick-wall Limiter with a fixed Lookahead of 10 ms adding the corresponding delay to the output signal.\n\n**Threshold** \nThis parameter represents the ideal amplification level needed by the signal.\n\nIt contributes along with the Ceiling to determine the amount of gain to apply to the signal before the limiting stage. The gain is calculated by `(-Threshold) - (-Ceiling)`.\n\nFor example, on Threshold -6 dB and Ceiling -2 dB, the signal is amplified by 4 dB and limited to not exceed -2 dB output level.\n\n**Ceiling** \nThis parameter represents the ideal attenuation level needed by the signal and the maximum allowed output level.\n\nWhen the Threshold is set to 0 dB, the Ceiling is simply the gain reduction. When the Threshold is lowered, the signal is boosted without overtaking the Ceiling value.\n\n**Release** \nSets the release of the internal brick-wall Limiter. Lower values may introduce small artifacts.\n\n## References\n\n- [Sonarworks Blog - What Is A Maximizer?](https://www.sonarworks.com/soundid-reference/blog/learn/what-is-a-maximizer/)\n","src/contents/docs/plugins/midside_equalizer.md":"# Mid-Side Equalizer\n\nThe Mid-Side Equalizer processes the Mid (sum) and Side (difference) parts of a stereo signal independently. It's useful, for example, to fix the overall tone of a recording through the Mid channel while keeping the stereo ambience untouched on the Side channel. Easy Effects uses the Parametric Equalizer x32 MidSide from Linux Studio Plugins. You can run anywhere from 1 to 32 bands per channel, and tweak the width and center frequency of each band.\n\n## Global Options\n\n**Bands** \nNumber of bands.\n\n**Mode** \n- **IIR** - Infinite Impulse Response filters, nonlinear minimal phase. Usually doesn't add noticeable latency to the output.\n- **FIR** - Finite Impulse Response filters with linear phase, a finite approximation of the equalizer's impulse response. Adds noticeable latency to the output.\n- **FFT** - Fast Fourier Transform approximation of the frequency chart, linear phase. Adds noticeable latency to the output.\n\n**Balance** \nBalance between the left and right output channels.\n\n**Pitch Mid** \nFrequency shift for all mid channel filters, in semitones.\n\n**Pitch Side** \nFrequency shift for all side channel filters, in semitones.\n\n**Link Mid/Side** \nWhen enabled, you can apply different configurations to the mid and side channels.\n\n**Flat Response** \nResets all band gains to 0.\n\n**Calculate Frequencies** \nFigures out the center frequency and width for each band based on the current band count. Handy if you want fewer than 32 bands but aren't sure which frequencies to pick.\n\n## Band Options\n\n**Type** \n- **Off** - Filter is disabled.\n- **Bell** - Bell filter with smooth peak/recess.\n- **High Pass** - High Pass filter that removes low frequencies.\n- **High Shelf** - Shelving filter that adjusts the high frequency range.\n- **Low Pass** - Low Pass filter that removes high frequencies.\n- **Low Shelf** - Shelving filter that adjusts the low frequency range.\n- **Notch** - Notch filter that fully rejects the selected frequency.\n- **Resonance** - Resonance filter with sharp peak/recess.\n- **All Pass** - All Pass filter.\n\n**Mode** \n- **RLC** - Very smooth filters built from cascaded RLC contours. Uses Bilinear Z-transform (BT) or Matched Z-transform (MT) for pole/zero mapping.\n- **BWC** - Butterworth-Chebyshev-type-1 based filters. Doesn't affect Resonance and Notch filters. Uses Bilinear Z-transform (BT) or Matched Z-transform (MT) for pole/zero mapping.\n- **LRX** - Linkwitz-Riley based filters. Doesn't affect Resonance and Notch filters. Uses Bilinear Z-transform (BT) or Matched Z-transform (MT) for pole/zero mapping.\n- **APO** - Digital biquad filters derived from canonical analog biquad prototypes digitalized through Bilinear transform. These are [textbook filters](https://shepazu.github.io/Audio-EQ-Cookbook/audio-eq-cookbook.html) implemented the same way as in [Equalizer APO](https://equalizerapo.com/). Direct design (DR) generates the digital filter coefficients directly in the digital domain, without any transforms.\n\n**Slope** \nSlope of the filter curve.\n\n**Solo** \nMakes the selected band the only active one.\n\n**Mute** \nMutes the selected band.\n\n**Frequency** \nCenter frequency of the band.\n\n**Width** \nBandwidth, calculated as `width = frequency / quality`.\n\n**Quality** \nQuality factor of the filter.\n\n## References\n\n- [Wikipedia Equalization (audio)](https://en.wikipedia.org/wiki/Equalization_(audio))\n- [LSP Parametric Equalizer x32 MidSide](http://lsp-plug.in/?page=manuals§ion=para_equalizer_x32_ms)\n- [Wikipedia Q Factor](https://en.wikipedia.org/wiki/Q_factor)\n- [How to EQ - Q Factor and Bandwidth in EQ: What They Mean](https://howtoeq.wordpress.com/2010/10/07/q-factor-and-bandwidth-in-eq-what-it-all-means/)\n","src/contents/docs/plugins/multibandcompressor.md":"# Multiband Compressor\n\nEasy Effects uses the Sidechain Multiband Compressor Stereo developed by Linux Studio Plugins. Please refer to the [Compressor](compressor.md) documentation to comprehend the basic functionality of the dynamic range compression.\n\n## Global Options\n\n**Band Management** \nThis Compressor allows to split the input signal up to 8 bands. Each band is not attached to its strict frequency and can be controlled by completely different frequency range that can be obtained by applying Low-Cut and Hi-Cut filters to the Sidechain signal.\n\nThe first band is always enabled while the others can be activated if needed. When only the first band is enabled, the functionality is similar to the Singleband Sidechain Compressor. Otherwise the signal is split in more bands and each band is compressed individually. After the compression stage, the bands are mixed together to form the output result.\n\n**Operating Mode** \nDetermines how the input signal is split to obtain the different bands.\n\n- **Classic** - The original signal is split using crossover filters. After the compression stage, all bands become phase-compensated using All-Pass filters.\n- **Modern** - Each band is processed by a pair of dynamic shelving filters. This mode allows a better control over the gain of each band.\n\n**Sidechain Boost** \nIf enabled, it introduces a special mode for assigning the same weight for higher frequencies opposite to lower frequencies.\n\nWhen disabled, the frequency band is processed 'as is', but the usual audio signal has 3 dB/octave falloff in the frequency domain and could be compared with the pink noise. So lower frequencies take more effect on the Compressor rather than higher frequencies.\n\nOn the other hand the Sidechain Boost mode allows to compensate the -3 dB/octave falloff of the signal spectrum and, even more, makes the signal spectrum growing +3 dB/octave in the almost fully audible frequency range.\n\n**Pink** applies +3 dB/octave while **Brown** applies +6 dB/octave Sidechain Boost. Each of them can use bilinear-transformed (BT) or matched-transformed (MT) shelving filter.\n\n**Sidechain Source** \nIf the External Sidechain is enabled inside at least one band, this combobox allows to select the input device as source.\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal mixed in the output.\n\n## Band Options\n\n**Band Start** \nAllows to change the lower end split frequency of the selected band. This value is assigned to the Band End of the previous enabled band. It can be modified for all bands except the first one, which is always enabled and its value is 0 Hz.\n\n**Band End** \nSpecify the upper end split frequency of the selected band. It cannot be directly modified and assumes the same value of the Band Start of the next enabled band. For the last enabled band it is always 24.000 Hz.\n\n**Compression Mode** \n- **Downward** - It's aimed to decrease the gain of the signal above the Threshold.\n- **Upward** - It's aimed to increase the gain of the signal below the Threshold.\n- **Boosting** - It's aimed to increase the gain of the signal below the Threshold by a specific amount.\n\n**External Sidechain** \nThe Sidechain is an external source took by a specific input device (typically a microphone).\n\n**Band Bypass** \nIf enabled, the selected band is not affected by the compression stage.\n\n**Solo** \nTurns on the Solo mode to the selected band by applying -36 dB gain to the other non-soloing bands.\n\n**Mute** \nTurns on the Mute mode applying -36 dB gain to to the selected band.\n\n**Attack Time** \nThe length of time it takes to apply roughly two-thirds of the targeted amount of compression ratio to the uncompressed band signal.\n\n**Release Time** \nThe length of time it takes to restore roughly two-thirds of the reduced gain (in Downward mode) or increased gain (in Upward/Boosting mode) to the compressed band signal.\n\n**Attack Threshold** \nThe target level around which the compression is applied (the range depends by the Knee).\n\n**Release Threshold** \nSets up the Threshold of the Release Time, calculated by summing the Release Threshold to the Attack Threshold.\n\nIf the Sidechain level is above that Threshold, the compressor uses the Release Time for the releasing stage. Otherwise the Attack Time is used in place of Release Time.\n\nFor example, with -10 dB Attack Threshold and -60 dB Release Threshold, if the Sidechain is above `-10 + (-60) = -70 dB`, the Release Time is used for the gain restoration. If the Sidechain is below -70 dB, the Attack Time is used for the gain restoration.\n\n**Ratio** \nThe amount of attenuation (in Downward mode) or amplification (in Upward/Boosting mode) that will be applied to the signal.\n\nFor example, when the Ratio is 2 in Downward mode and the Sidechain rises above the Threshold by 10 dB, the signal would be ideally reduced by 5 dB (10/2 dB). In practice this behavior mostly depends on how the Multiband Compressor is designed and configured.\n\n**Knee** \nThe range over which the Compressor switches from no compression to almost the full ratio compression (the Threshold usually sits at the center of this transition zone).\n\n**Makeup** \nThe gain to apply after the compression stage.\n\n## Band Sidechain Options\n\n**Mode** \nDetermines how the Sidechain of the selected band is evaluated for the compression stage.\n\n- **Peak** - The Compressor reacts according to the peaks.\n- **RMS** - The Compressor reacts according to the average loudness measured by the root mean square.\n- **Low-Pass** - The Compressor reacts according to the signal processed by a Low-Pass filter.\n- **Uniform** - The Compressor reacts according to the loudness measured by the average of the absolute amplitude.\n\n**Source** \nDetermines which part of the Sidechain is taken into account for the compression stage.\n\n- **Middle** - The sum of left and right channels.\n- **Side** - The difference between left and right channels.\n- **Left** - Only left channel is used.\n- **Right** - Only right channel is used.\n- **Min** - The absolute minimum value is taken from stereo input.\n- **Max** - The absolute maximum value is taken from stereo input.\n\n**Low-Cut Filter** \nEnables a custom Low-Cut Filter for the selected band.\n\n**Low-Cut Frequency** \nSets the cut-off frequency of the custom Low-Cut filter. If it is disabled, the default Low-Cut filter assumes internally the Band Start frequency as the cut-off frequency.\n\n**Hight-Cut Filter** \nEnables a custom High-Cut Filter for the selected band.\n\n**Hight-Cut Frequency** \nSets the cut-off frequency of the custom High-Cut filter. If it is disabled, the default High-Cut filter assumes internally the Band End frequency as the cut-off frequency.\n\n**PreAmp** \nGain applied to the Sidechain signal of the selected band.\n\n**Reactivity** \nThe time that defines the number of samples used to process the Sidechain in RMS, Uniform and Low-Pass modes. Higher the value, more smooth the compression.\n\n**Lookahead** \nThe band signal to compress is delayed by this amount of time, so that the compression will be applied earlier than it would be otherwise.\n\nEach band can have different Lookahead values. To avoid phase distortions in the mixing stage, all the bands are automatically delayed for an individually calculated period of time.\n\n**Boost Amount** \nMaximum gain amplification to apply in Boosting Mode.\n\n**Boost Threshold** \nThe Threshold below which a constant amplification will be applied to the band signal in Upward Mode (it prevents from applying infinite amplification to very quiet signals).\n\n## References\n\n- [Wikipedia Dynamic Range Compression](https://en.wikipedia.org/wiki/Dynamic_range_compression)\n- [LSP Sidechain Multiband Compressor Stereo](https://lsp-plug.in/?page=manuals§ion=sc_mb_compressor_stereo)\n- [Black Ghost Audio - The Ultimate Guide to Compression](https://www.blackghostaudio.com/blog/the-ultimate-guide-to-compression)\n- [Attack Magazine - Demolishing The Myths of Compression](https://www.attackmagazine.com/features/columns/gregory-scott-demolishing-the-myths-of-compression/)\n","src/contents/docs/plugins/multibandgate.md":"# Multiband Gate\n\nEasy Effects uses the Multiband Gate Stereo developed by Calf Studio Gear. Please refer to the [Gate](gate.md) documentation to comprehend the basic functionality of the gating process.\n\n## Global Options\n\n**Band Management** \nThis Gate allows to split the input signal up to 8 bands. Each band is not attached to its strict frequency and can be controlled by completely different frequency range that can be obtained by applying Low-Cut and Hi-Cut filters to the Sidechain signal.\n\nThe first band is always enabled while the others can be activated if needed. When only the first band is enabled, the functionality is similar to the Singleband Sidechain Gate. Otherwise the signal is split in more bands and each band is processed individually. After the gating stage, the bands are mixed together to form the output result.\n\n**Operating Mode** \nDetermines how the input signal is split to obtain the different bands.\n\n- **Classic** - The original signal is split using crossover filters. After the gating stage, all bands become phase-compensated using All-Pass filters.\n- **Modern** - Each band is processed by a pair of dynamic shelving filters. This mode allows a better control over the gain of each band.\n\n**Sidechain Boost** \nIf enabled, it introduces a special mode for assigning the same weight for higher frequencies opposite to lower frequencies.\n\nWhen disabled, the frequency band is processed 'as is', but the usual audio signal has 3 dB/octave falloff in the frequency domain and could be compared with the pink noise. So lower frequencies take more effect on the Gate rather than higher frequencies.\n\nOn the other hand the Sidechain Boost mode allows to compensate the -3 dB/octave falloff of the signal spectrum and, even more, makes the signal spectrum growing +3 dB/octave in the almost fully audible frequency range.\n\n**Pink** applies +3 dB/octave while **Brown** applies +6 dB/octave Sidechain Boost. Each of them can use bilinear-transformed (BT) or matched-transformed (MT) shelving filter.\n\n**Sidechain Source** \nIf the External Sidechain is enabled inside at least one band, this combobox allows to select the input device as source.\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal mixed in the output.\n\n## Band Options\n\n**Band Start** \nAllows to change the lower end split frequency of the selected band. This value is assigned to the Band End of the previous enabled band. It can be modified for all bands except the first one, which is always enabled and its value is 0 Hz.\n\n**Band End** \nSpecify the upper end split frequency of the selected band. It cannot be directly modified and assumes the same value of the Band Start of the next enabled band. For the last enabled band it is always 24.000 Hz.\n\n**External Sidechain** \nThe Sidechain is an external source took by a specific input device (typically a microphone).\n\n**Band Bypass** \nIf enabled, the selected band is not affected by the gating stage.\n\n**Solo** \nTurns on the Solo mode to the selected band by applying -36 dB gain to the other non-soloing bands.\n\n**Mute** \nTurns on the Mute mode applying -36 dB gain to to the selected band.\n\n**Attack Time** \nThe length of time it takes to apply roughly two-thirds of the gain reduction.\n\n**Release Time** \nThe length of time it takes to restore roughly two-thirds of the gain reduction.\n\n**Curve Threshold** \nThe Gate fully opens upon the Sidechain level becoming above Curve Threshold.\n\nIf Hysteresis is not enabled, the Gate begins to close upon the Sidechain level becoming below Curve Threshold.\n\n**Curve Zone Size** \nThe Gate begins to open upon the Sidechain level becoming above the Curve Threshold + Curve Zone.\n\nIf Hysteresis is not enabled, the Gate fully closes upon the Sidechain level becoming below the Curve Threshold + Curve Zone.\n\n**Hysteresis** \nWhen enabled, Curve Threshold and Curve Zone apply only to the opening Gate, and separate parameters can be configured for closing Gate.\n\n**Hysteresis Threshold** \nIf Hysteresis is enabled, the Gate begins to close upon the Sidechain level becoming below Curve Threshold + Hysteresis Threshold.\n\n**Hysteresis Zone Size** \nIf Hysteresis is enabled, the Gate fully closes upon the Sidechain level becoming below the Curve Threshold + Hysteresis Threshold + Hysteresis Zone.\n\n**Reduction** \nIf the value is negative, it acts as the amount of gain reduction to apply to the input signal when the Gate is fully closed. If the value is positive, the Gate operates in \"Reverse Mode\": It reduces the amplitude when the signal is above the threshold.\n\n**Makeup** \nThe gain to apply after the gating stage.\n\n## Band Sidechain Options\n\n**Mode** \nDetermines how the Sidechain is evaluated for the gating stage.\n\n- **Peak** - The Gate reacts according to the peaks.\n- **RMS** - The Gate reacts according to the average loudness measured by the root mean square.\n- **Low-Pass Filter** - The Gate reacts according to the signal processed by a recursive 1-pole Low-Pass filter.\n- **Simple Moving Average** - The Gate reacts according to the signal processed by the Simple Moving Average filter.\n\n**Source** \nDetermines which part of the Sidechain is taken into account for the gating stage.\n\n- **Middle** - The sum of left and right channels.\n- **Side** - The difference between left and right channels.\n- **Left** - Only left channel is used.\n- **Right** - Only right channel is used.\n- **Min** - The absolute minimum value is taken from stereo input.\n- **Max** - The absolute maximum value is taken from stereo input.\n\n**Low-Cut Filter** \nEnables a custom Low-Cut Filter for the selected band.\n\n**Low-Cut Frequency** \nSets the cut-off frequency of the custom Low-Cut filter. If it is disabled, the default Low-Cut filter assumes internally the Band Start frequency as the cut-off frequency.\n\n**Hight-Cut Filter** \nEnables a custom High-Cut Filter for the selected band.\n\n**Hight-Cut Frequency** \nSets the cut-off frequency of the custom High-Cut filter. If it is disabled, the default High-Cut filter assumes internally the Band End frequency as the cut-off frequency.\n\n**PreAmp** \nGain applied to the Sidechain signal of the selected band.\n\n**Reactivity** \nThe time that defines the number of samples used to process the Sidechain in RMS, Uniform and Low-Pass modes. Higher the value, more smooth the gating.\n\n**Lookahead** \nThe band signal to gate is delayed by this amount of time, so that the gating will be applied earlier than it would be otherwise.\n\nEach band can have different Lookahead values. To avoid phase distortions in the mixing stage, all the bands are automatically delayed for an individually calculated period of time.\n\n## References\n\n- [LSP Sidechain Multiband Gate Stereo](https://lsp-plug.in/?page=manuals§ion=sc_mb_gate_stereo)\n- [Wikipedia Noise Gate](https://en.wikipedia.org/wiki/Noise_gate)\n","src/contents/docs/plugins/pitch.md":"# Pitch\n\nPitch shifting is a sound recording technique in which the original Pitch of a sound is raised or lowered. Easy Effects uses the pitch shifter from SoundTouch.\n\n**Mode** \nControls the method used for pitch shifting.\n\n- **High Speed** - Uses a method with a CPU cost that is relatively moderate and predictable.\n- **High Quality** - Uses the highest quality method for pitch shifting. This CPU cost is approximately proportional to the required frequency shift.\n- **High Consistency** - Uses the method that gives greatest consistency when used to create small variations in pitch around the 1.0-ratio level. Unlike the previous two options, this avoids discontinuities when moving across the 1.0 pitch scale. It also consumes more CPU than the others in the case where the pitch scale is exactly 1.0.\n\n**Formant** \nControls the handling of formant shape (spectral envelope) when pitch-shifting.\n\n- **Shifted** - Applies no special formant processing. The spectral envelope will be pitch shifted as normal.\n- **Preserved** - Preserves the spectral envelope of the unshifted signal. This permits shifting the note frequency without so substantially affecting the perceived pitch profile of the voice or instrument.\n\n**Transients** \nControls the component frequency phase-reset mechanism that may be used at transient points to provide clarity and realism to percussion and other significant transient sounds.\n\n- **Crisp** - Resets component phases at the peak of each transient (the start of a significant note or percussive event). This usually results in a clear-sounding output, but it is not always consistent, and may cause interruptions in stable sounds present at the same time as transient events.\n- **Mixed** - Resets component phases at the peak of each transient, outside a frequency range typical of musical fundamental frequencies. The results may be more regular for mixed stable and percussive notes than Crisp option, but with a \"phasier\" sound. The balance may sound very good for certain types of music and fairly bad for others.\n- **Smooth** - Does not reset component phases at any point. The results will be smoother and more regular but may be less clear than the other transient options.\n\n**Detector** \nControls the type of transient detector used.\n\n- **Compound** - Uses a general purpose transient detector which is likely to be good for most situations.\n- **Percussive** - Detects percussive transients.\n- **Soft** - Uses an onset detector with less of a bias toward percussive transients. This may give better results with certain material (e.g. relatively monophonic piano music).\n\n**Phase** \nControls the adjustment of component frequency phases from one analysis window to the next during non-transient segments.\n\n- **Laminar** - Adjusts phases when stretching in such a way as to try to retain the continuity of phase relationships between adjacent frequency bins whose phases are behaving in similar ways. This should give good results in most situations.\n- **Independent** - Adjusts the phase in each frequency bin independently from its neighbours. This usually results in a slightly softer, phasier sound.\n\n**Cents** \nNumber of cents the Pitch will be increased or decreased.\n\n**Semitones** \nNumber of semitones the Pitch will be increased or decreased.\n\n**Octaves** \nNumber of octaves the Pitch will be increased or decreased.\n\n## References\n\n- [Wikipedia Pitch Shift](https://en.wikipedia.org/wiki/Pitch_shift)\n- [SoundTouch Audio Time Stretcher Library - Attractive Features](https://www.surina.net/soundtouch/)\n","src/contents/docs/plugins/reverb.md":"# Reverberation\n\nReverberation is the phenomenon of persistence of sound after the source has been stopped as a result of multiple reflections of the waves over objects within a closed surface. These reflections build up with each other and decay gradually as they are absorbed by the surfaces of objects in the space enclosed.\n\nThe Reverberation is different than Echo because the Echo is a reflected sound wave with sufficient magnitude and delay to be detectable as a signal distinct from the source one. To simulate the Reverberation effect, Easy Effects uses the Reverb plugin developed by Calf Studio Gear.\n\n**High Frequency Damping** \nCutoff frequency of the reflections. It causes higher frequencies to decay faster.\n\n**Room Size** \nSize of the space where simulated Reverberation occurs inside. It determines the time between reflections.\n\n**Diffusion** \nDegree of uniformity. Higher values lead to less uniform Reverberation.\n\n**Pre Delay** \nAdditional delay. It corresponds to a distance between sound source and the nearest wall.\n\n**Decay Time** \nThe time it takes for Reverberation to fade out.\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal (Reverberation) mixed in the output.\n\n**Bass Cut** \nRemoves low frequencies from the Reverberation.\n\n**Treble Cut** \nRemoves high frequencies from the Reverberation.\n\n## References\n\n- [Wikipedia Reverberation](https://en.wikipedia.org/wiki/Reverberation)\n- [Calf Reverb](https://calf-studio-gear.org/doc/Reverb.html)\n","src/contents/docs/plugins/rnnoise.md":"# Noise Reduction\n\nThe Noise Reduction is a process aimed to attenuate the disturbing noise from a signal.\n\nEasy Effects Noise Reduction is made on the RNNoise library which is based based on recurrent neural network, a class of artificial neural networks where connections between nodes form a directed graph along a temporal sequence. This allows it to exhibit temporal dynamic behavior.\n\nStandard RNNoise Model is used and custom models can be imported to perform different types of noise reduction.\n\n## References\n\n- [Wikipedia Noise Reduction](https://en.wikipedia.org/wiki/Noise_reduction)\n- [Wikipedia Recurrent Neural Network](https://en.wikipedia.org/wiki/Recurrent_neural_network)\n- [Jean-Marc Valin - RNNoise: Learning Noise Suppression](https://jmvalin.ca/demo/rnnoise/)\n","src/contents/docs/plugins/spectrum.md":"# Spectrum\n\n**Enabled** \nShow or hide the Spectrum.\n\n**Shape** \nChoose the style of the animation.\n\n**Points** \nNumber of points/bars on the x axis.\n\n**Height** \nChanges the height of the Spectrum.\n\n**Line Width** \nThe width of a single bar.\n\n**Fill** \nDraw filled bars.\n\n**Show Bars Border** \nDraw bars with borders.\n\n**Rounded Corners** \nDraw bars with rounded corners.\n\n**Color** \nChoose the color of the Spectrum components.\n\n**Minimum Frequency Range** \nLower end frequency of the Spectrum.\n\n**Maximum Frequency Range** \nUpper end frequency of the Spectrum.\n\n[Settings Menu](../user_interface/settingsmenu.md)","src/contents/docs/plugins/speex.md":"# Speech Processor\n\nThis plugin allows EasyEffects to use the Speex preprocessor to attenuate disturbing background noises from a signal.\n\nCompared to Noise Reduction which uses RNNoise to suppress noises, Speech Processor has the benefit of using less computational resources, at the cost of sacrificing noise suppression quality.\n\nFor more information on noise suppression in general, refer to the manual page on Noise Reduction.\n\n## References\n\n- [The Speex Project](https://www.speex.org/)\n","src/contents/docs/plugins/stereotools.md":"# Stereo Tools\n\nEasy Effects uses the Stereo Tools developed by Calf Studio Gear which provides some handy utilities to manage Stereo streams handling Left and Right channels in conjunction with Mid and Side signals.\n\nMid/Side is a recording technique which registers Mid and Side signals rather then plain Left and Right channels. Its basic setup makes use of one cardioid microphone (Mid) and one bidirectional (figure-eight) microphone (Sides). In mastering stage Mid and Side can be used to shape the balance of the Stereo image having more control over the mix to get a wider, deeper, and more focused track.\n\nHaving a Stereo stream, Mid is obtained summing both channels `(L+R)` while Side is the result of the difference between Left and Right `(L-R)`. Mid and Side signals can be decoded back to Left `(M+S)/2` and Right `(M-S)/2`.\n\n## Input\n\n**Balance** \nSets the balance between both channels.\n\n**Softclip (button)** \nMakes a kind of analog distortion instead of harsh digital 0 dB clipping.\n\n**Softclip (value)** \nLevel of Softclip.\n\n## Stereo Matrix\n\n**Mode** \nAllows to choose between different channel configurations such as normal Stereo Mode, Mid-Side or Mono simulation.\n\n**Mute L/R** \nMutes the Left/Right channel.\n\n**Invert Phase L/R** \nChanges the Phase of the Left/Right channel.\n\n**Side Level** \nThe level of the Side signal.\n\n**Side Balance** \nThe balance of the Side signal.\n\n**Middle Level** \nThe level of the Middle signal.\n\n**Middle Panorama** \nThe position in the Panorama of the Middle signal.\n\n## Output\n\n**Balance** \nSets the balance between both channels.\n\n**Delay L/R** \nDelays the Left or the Right channel. Negative values delay the Left channel and positive values the Right channel.\n\n**Stereo Base** \nSets the Stereo Base of the content seamless between Mono and inverted channels.\n\n**Stereo Phase** \nSet the Stereo Phase of the content.\n\n## References\n\n- [Wikipedia Stereophonic Sound](https://en.wikipedia.org/wiki/Stereophonic_sound)\n- [Calf Stereo Tools](https://calf-studio-gear.org/doc/Stereo%20Tools.html)\n- [Unlock Your Sound - Stereo, Mono, Mid, Side, Panning, and Imaging Explained](https://unlockyoursound.com/stereo/)\n","src/contents/docs/plugins/voicesuppressor.md":"# Voice Suppressor\n\nVery simple plugin that removes voice from audio. It calculates the correlation of the lefft and right channels in\nthe frequency domain. Frequencies whose correlation and phase difference are beyond the configured thresholds are attenuated. \n\n**Start** \n\nFrequencies below this value are not attenuated. \n\n**End** \n\nFrequencies above this value value are not attenuated. \n\n**Correlation** \n\nFrequencies are attenuated only if the correlation between left and right channel is at least equal to this value. \n\n**Phase Difference** \n\nFrequencies are not attenuated when the magnitude of phase difference between left and right channel is above this value. \n\n**Minimum Kurtosis** \n\nFrequencies with local kurtosis below this value are attenuated. \n\n**Maximum Instantaneous Frequency** \n\nThis is related to the frequency of rotation of the complex vector that represents the correlation between the left and right channel at a given frequency.\nIn other words this measures how fast the phase difference between the left and right channels changes at a given frequency bin in the Fourier transform.\nMore attenuation is applied to the channels samples that have rotation frequency below this parameter value. \n\n**Inverted Mode** \nInstead of suppressing voice the plugin will try to suppress the background and keep the voice. \n"},"items":[{"name":"autogain.md","path":"src/contents/docs/plugins/autogain.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/autogain.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Auto Gain\n\nEasy Effects Autogain is based on the library libebur128 which implements the EBU R 128 standard for loudness normalization. It changes the audio volume to a perceived loudness target that can be customized by the user.\n\n**Target** \nLoudness level.\n\n**Silence** \nSilence threshold. While the momentary loudness is below this value the autogain won't make any changes to the current gain being applied to the signal level.\n\n**Maximum History** \nRange of time taken into account for the calculation of loudness level and output gain.\n\n**Reference** \nThe parameter used as reference to evaluate the output gain.\n\n- **Momentary** - Measures the loudness of the past 400 milliseconds.\n- **Short-Term** - Measures the loudness of the past 3 seconds.\n- **Integrated** - Ideally it indicates how loud the content is on average. It measures the loudness on a long range of time, depending on the value set as Maximum History.\n- **Geometric Mean** - Uses the geometric mean of all the above mentioned parameters, or two of them at user choosing.\n\n**Reset History** \nResets the Autogain history related to chosen Reference.\n\n**Monitor Parameters** \nAutogain values shown as stats.\n\n- **Relative** - Used to detect silence. Whenever the Momentary term is below a predetermined threshold, modifications to the output gain will be disabled.\n- **Loudness** - The difference between its value and the target loudness determines the output gain.\n- **Range** - Indicates how large is the dynamic range of the content played.\n- **Output Gain** - The input signal is adjusted by this correction gain to bring its loudness to the target value.\n\n## References\n\n- [Wikipedia EBU R 128](https://en.wikipedia.org/wiki/EBU_R_128)\n- [EBU - Loudness Normalisation and Permitted Maximum Level of Audio Signals](https://tech.ebu.ch/publications/r128/)\n","isInternal":false,"tokens":413,"sizeBytes":1841},{"name":"autotune.md","path":"src/contents/docs/plugins/autotune.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/autotune.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Autotune\n\nAutotune is a pitch correction effect that automatically adjusts the pitch of an audio signal to the nearest target note. It can be used to subtly correct vocal intonation or to create the characteristic \"auto-tune effect\" heard in modern music. Easy Effects uses the fat1 plugin from x42 LV2 Plugins.\n\n## Tuning Options\n\n**Mode** \nControls how pitch correction is applied.\n\n- **Auto** - Automatically detects the pitch and corrects it to the nearest enabled note in the scale.\n- **Manual** - Targets a fixed pitch set by the Offset parameter.\n\n**Tuning** \nThe reference frequency for the A4 note, in Hz. The standard tuning is 440 Hz. Adjust this if you are working with non-standard tuning (range: 400–480 Hz).\n\n**Correction** \nControls the strength of pitch correction. At 1.0, the pitch is fully corrected to the target note. Lower values allow more of the natural pitch variation through, producing a subtler effect.\n\n**Bias** \nControls the preference toward the current note versus the nearest target note. At 0.5 (center), no bias is applied. Higher values make the correction stick to the current note longer before switching. Lower values make it switch to the nearest target more readily.\n\n**Filter** \nControls the smoothness of the pitch detection. Lower values result in faster, more responsive detection but may introduce artifacts. Higher values produce smoother, more stable detection but with slower response.\n\n**Offset** \nShifts the target pitch by the specified number of semitones (range: -2.0 to 2.0). In Manual mode, this sets the fixed target pitch offset.\n\n**Fast Correction** \nWhen enabled, applies pitch correction with minimal latency. This produces a more immediate, aggressive correction effect but may introduce more artifacts.\n\n## Scale Selection\n\nThe Scale section allows you to select which notes the pitch correction will target.\n\n**Root** \nThe root note of the scale (C, D, E, F, G, A, or B).\n\n**Accidental** \nApplies a sharp or flat modifier to the root note.\n\n**Scale** \nSelects either Major or Minor scale pattern. Choosing a scale automatically enables the correct notes for that key.\n\n**Note Toggles** \nIndividual toggles for each of the 12 chromatic notes (C through B). Selecting a scale sets these automatically, but you can also configure them manually for custom scales or chromatic correction (all notes enabled).\n\n## Pitch Error Meter\n\nDisplays the current pitch deviation from the target note. A value of 0 means the pitch is exactly on target. Negative values indicate the pitch is flat, positive values indicate it is sharp.\n\n## References\n\n- [x42 fat1 Auto Tune](https://x42-plugins.com/x42/x42-autotune)\n- [Wikipedia Auto-Tune](https://en.wikipedia.org/wiki/Auto-Tune)\n","isInternal":false,"tokens":602,"sizeBytes":2745},{"name":"bassenhancer.md","path":"src/contents/docs/plugins/bassenhancer.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/bassenhancer.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Bass Enhancer\n\nA Bass Enhancer is used to produce very low sound that is not present in the original signal. This is done by creating harmonic distortions of the signal which are restricted in range and added to the original signal. It raises the lower end of an audio signal without simply raising the lower frequencies like an [Equalizer](equalizer.md) would do to create a more \"fat\" or \"boomy\" sound.\n\n**Amount** \nAmount of harmonics added to the original signal.\n\n**Harmonics** \nAmount of newly created harmonics.\n\n**Scope** \nThe frequency above which harmonics are produced.\n\n**Floor (button)** \nConstrain the enhancement on the lower end.\n\n**Floor (value)** \nThe frequency below which no harmonics are produced.\n\n**Blend Harmonics** \nThe \"colour\" (or octave) of the harmonics.\n\n**Listen** \nMute the original signal and listen to the harmonics exclusively.\n\n## References\n\n- [Calf Bass Enhancer](https://calf-studio-gear.org/doc/Bass%20Enhancer.html)\n- [Wikipedia Missing Fundamental](https://en.wikipedia.org/wiki/Missing_fundamental)\n","isInternal":false,"tokens":244,"sizeBytes":1051},{"name":"bassloudness.md","path":"src/contents/docs/plugins/bassloudness.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/bassloudness.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Bass Loudness\n\nThe ear is less sensitive to low frequencies when listening at low volume. This plugin developed by MDA allows the bass level to be adjusted measuring the \"equal-loudness contour\".\n\n**Loudness** \nSource level relative to listening level (based on a 100 dB SPL maximum level).\n\n**Output** \nChange output level.\n\n**Link** \nAutomatically adjusts Output to maintain a consistent tonal balance at all levels.\n\n## References\n\n- [Wikipedia Equal-Loudness Contour](https://en.wikipedia.org/wiki/Equal-loudness_contour)\n- [MDA Loudness](http://mda.smartelectronix.com/vst/help/loudness.htm)\n","isInternal":false,"tokens":142,"sizeBytes":602},{"name":"compressor.md","path":"src/contents/docs/plugins/compressor.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/compressor.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Compressor\n\nA Compressor is used to reduce the dynamic range or, in other words, the difference in level between the quietest and the loudest parts of an audio signal. It achieves this purpose altering the gain when the signal overtakes a predetermined Threshold. Easy Effects uses the Stereo Sidechain Compressor from Linux Studio Plugins.\n\n## Compressor Options\n\n**Attack Time** \nThe length of time it takes to apply roughly two-thirds of the targeted amount of compression ratio to the uncompressed signal.\n\n**Release Time** \nThe length of time it takes to restore roughly two-thirds of the reduced gain (in Downward mode) or increased gain (in Upward/Boosting mode) to the compressed signal.\n\n**Attack Threshold** \nThe target level around which the compression is applied (the range depends by the Knee).\n\n**Release Threshold** \nSets up the Threshold of the Release Time, calculated by summing the Release Threshold to the Attack Threshold.\n\nIf the Sidechain level is above that Threshold, the compressor uses the Release Time for the releasing stage. Otherwise the Attack Time is used in place of Release Time.\n\nFor example, with -10 dB Attack Threshold and -60 dB Release Threshold, if the Sidechain is above `-10 + (-60) = -70 dB`, the Release Time is used for the gain restoration. If the Sidechain is below -70 dB, the Attack Time is used for the gain restoration.\n\n**Ratio** \nThe amount of attenuation (in Downward mode) or amplification (in Upward/Boosting mode) that will be applied to the signal.\n\nFor example, when the Ratio is 2 in Downward mode and the Sidechain rises above the Threshold by 10 dB, the signal would be ideally reduced by 5 dB (10/2 dB). In practice this behavior mostly depends on how the Compressor is designed and configured.\n\n**Knee** \nThe range over which the Compressor switches from no compression to almost the full ratio compression (the Threshold usually sits at the center of this transition zone).\n\n**Makeup** \nThe gain to apply after the compression stage.\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal mixed in the output.\n\n**Mode** \n- **Downward** - It's aimed to decrease the gain of the signal above the Threshold.\n- **Upward** - It's aimed to increase the gain of the signal below the Threshold.\n- **Boosting** - It's aimed to increase the gain of the signal below the Threshold by a specific amount.\n\n**Boost Threshold** \nThe Threshold below which a constant amplification will be applied to the input signal in Upward Mode (it prevents from applying infinite amplification to very quiet signals).\n\n**Boost Amount** \nMaximum gain amplification to apply in Boosting Mode.\n\n## Sidechain\n\n**Listen** \nAllows to listen the processed Sidechain signal.\n\n**Input Type** \nDetermines which signal is the Sidechain or, in other words, the signal that controls the compression stage.\n\n- **Feed-forward** - The Sidechain is the Compressor input signal (taken after applying the plugin input gain). More aggressive compression.\n- **Feed-back** - The Sidechain is the Compressor output signal (taken before applying the Makeup and the plugin output gain). Vintage-style compression.\n- **External** - The Sidechain is an external source took by a specific input device (typically a microphone).\n\n**Input Device** \nSelect the device for the External Sidechain.\n\n**Mode** \nDetermines how the Sidechain is evaluated for the compression stage.\n\n- **Peak** - The Compressor reacts according to the peaks.\n- **RMS** - The Compressor reacts according to the average loudness measured by the root mean square.\n- **Low-Pass** - The Compressor reacts according to the signal processed by a Low-Pass filter.\n- **Uniform** - The Compressor reacts according to the loudness measured by the average of the absolute amplitude.\n\n**Source** \nDetermines which part of the Sidechain is taken into account for the compression stage.\n\n- **Middle** - The sum of left and right channels.\n- **Side** - The difference between left and right channels.\n- **Left** - Only left channel is used.\n- **Right** - Only right channel is used.\n- **Min** - The absolute minimum value is taken from stereo input.\n- **Max** - The absolute maximum value is taken from stereo input.\n\n**PreAmplification** \nGain applied to the Sidechain signal.\n\n**Reactivity** \nThe time that defines the number of samples used to process the Sidechain in RMS, Uniform and Low-Pass modes. Higher the value, more smooth the compression.\n\n**Lookahead** \nThe signal to compress is delayed by this amount of time, so that the compression will be applied earlier than it would be otherwise. The corresponding delay is reproduced on the output signal.\n\n## Sidechain Filters\n\n**High-Pass Filter Mode** \nSets the type of the High-Pass filter applied to Sidechain signal.\n\n**High-Pass Frequency** \nSets the cut-off frequency of the High-Pass filter.\n\n**Low-Pass Filter Mode** \nSets the type of the Low-Pass filter applied to Sidechain signal.\n\n**Low-Pass Frequency** \nSets the cut-off frequency of the Low-Pass filter.\n\n## References\n\n- [Wikipedia Dynamic Range Compression](https://en.wikipedia.org/wiki/Dynamic_range_compression)\n- [LSP Sidechain Compressor Stereo](https://lsp-plug.in/?page=manuals§ion=sc_compressor_stereo)\n- [Black Ghost Audio - The Ultimate Guide to Compression](https://www.blackghostaudio.com/blog/the-ultimate-guide-to-compression)\n- [Attack Magazine - Demolishing The Myths of Compression](https://www.attackmagazine.com/features/columns/gregory-scott-demolishing-the-myths-of-compression/)\n","isInternal":false,"tokens":1222,"sizeBytes":5579},{"name":"convolver.md","path":"src/contents/docs/plugins/convolver.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/convolver.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Convolver\n\nThe Convolver creates a simulation of an audio environment using a pre-recorded audio sample of the impulse response of the space being modeled. This feature is based on the \"convolution\": a process through which the sonic characteristics of one signal are used to alter the character of another.\n\nEasy Effects Convolver offers the opportunity to apply multiple impulse responses by combining them in one file.\n\n**Impulses** \nManage impulse response files. This menu allows to load a specific impulse or add/remove files from the Easy Effects configuration directory.\n\n**Combine** \nSelect two impulse responses to combine into a single file to be saved under the Easy Effects configuration directory.\n\n**Stereo Width** \nModify the impulse response stereo image width.\n\n**Spectrum** \nVisualize the frequency spectrum of the selected channel.\n\n## References\n\n- [Wikipedia Reverb Effect](https://en.wikipedia.org/wiki/Reverb_effect)\n- [Indiana University - Convolution: A Form of Cross-Synthesis](https://cmtext.indiana.edu/synthesis/chapter4_convolution.php)\n- [Designing Sound - Recording Impulse Responses](https://designingsound.org/2012/12/29/recording-impulse-responses/)\n","isInternal":false,"tokens":246,"sizeBytes":1192},{"name":"crossfeed.md","path":"src/contents/docs/plugins/crossfeed.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/crossfeed.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Crossfeed\n\nEasy Effects uses the Crossfeed by bs2b. This plugin is used to improve headphone listening of stereo audio records. It does so by mixing the left and right channel in a way that simulates a stereo speaker setup while using headphones.\n\n**Cutoff** \nLow-Pass filter cutoff frequency.\n\n**Feed** \nAmount of signal from a channel that is sent to the other.\n\n**Presets** \n- **Default** - Closest to virtual speaker placement (30°, 3 meters).\n- **Cmoy** - Close to Chu Moy's Crossfeed (popular).\n- **Jmeier** - Close to Jan Meier's CORDA amplifiers (little change).\n\n## References\n\n- [Wikipedia Crossfeed](https://en.wikipedia.org/wiki/Crossfeed)\n- [Bauer Stereophonic-to-Binaural DSP.](http://bs2b.sourceforge.net/)\n","isInternal":false,"tokens":187,"sizeBytes":728},{"name":"crosstalkcanceller.md","path":"src/contents/docs/plugins/crosstalkcanceller.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/crosstalkcanceller.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Crosstalk Canceller\n\nWhen audio is played by a stereo speaker audio from both channels arrive to the listener ears. As the superposition\nof each channel audio will happen with a delay that depends on the distance between the listener ear and the speakers\nundesirable effects can happen. That superposition is called crosstalk in this context and can be attenuated by this\nplugin.","isInternal":false,"tokens":79,"sizeBytes":381},{"name":"crusher.md","path":"src/contents/docs/plugins/crusher.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/crusher.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Crusher\n\nA bitcrusher reduces the resolution or bandwidth of digital audio data. Audio reduced in bit depth sounds more harsh and \"digital\". \nIn the bitcrusher from Calf Studio Gear used in EasyEffects reduction can be done in a linear or logarithmic way.\nAccording to the plugin authors the logarithmic way results in a much smoother sound in low volume signals. \n\n**Mode** \n- **Linear** - Linear distance between bits.\n- **Logarithmic** - Logarithmic distances between bits. The result is a much more \"natural\" sounding crusher which doesn't gate low signals. \n\n**Bit reduction** \nControls the processed audio bit depth. \n\n**DC offset** \nThis offset causes different crushing of the lower and the upper half of the signal. \n\n**Anti-aliasing** \nControls the softness of the crushing sounds. \n\n**Mix** \nControls the mix of original and processed audio. When at 0% the output signal is equal to the original. When at 100% only the processed signal is in the output signal. \n\n**Reduction** \nControls the reduction of the sample rate (downsampling). \n\n**Low frequency oscillator active** \nEnable the low frequency oscillator. The oscillator will make the sample rate reduction change between a range determined by the range control.\n\n**Low frequency oscillator range** \nControls the amount of modulation applied to the sample rate reduction. \n\n**Low frequency oscillator rate** \nControls the frequency of the oscillator. \n\n## References\n\n- [Wikipedia Bit Crusher](https://en.wikipedia.org/wiki/Bitcrusher)\n- [Calf Crusher](https://calf-studio-gear.org)\n- [Calf Crusher Documentation](https://calf-studio-gear.org/doc/Crusher.html)\n","isInternal":false,"tokens":361,"sizeBytes":1640},{"name":"crystalizer.md","path":"src/contents/docs/plugins/crystalizer.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/crystalizer.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Crystalizer\n\nThe Crystalizer plugin can be used to add a little of dynamic range to songs that were overly compressed. The signal is split in multiple bands to which different intensities can be applied in order to alter the overall dynamic range.\n\n**Intensities**\nThe higher the value the higher is the difference in magnitude between the loudest and the quietest sounds of the selected band. Different intensities can be set for each frequency band.\n\n**Bypass**\nWhen active the audio signal passing through the selected band is not modified.\n\n**Mute**\nMutes the selected band.\n\n**Adaptive Intensity**\nWhen enabled the intensity value set by the band slider is continuously scaled as audio is processed. The scaling may\nincrease or decrease the intensity based on the geometric mean of the signal crest factor, kurtosis and spectral flux.\n\nWhen using static intensity values it may happen that the crystalizer enhances high frequency audio too much. What leads\nto audible artifacts. The adaptive scaling removes this is most of the cases.\n\n**Oversampling**\nResamples the signal to double its sampling rate value and back to the original value after the plugins effects are applied.\nIt improves the effects quality but it also increases CPU usage.\n\n**Oversampling Quality**\nControls the resampling quality. Higher values require more CPU power.\n\n**Fixed Quantum**\nForces the plugin to use a fixed audio buffer size to process audio. This helps to avoid noises when the sound server switches latency on the fly.\nBut it has the downside of introducing unneeded latency in some cases. And increased CPU usage in cases where higher latency could be used.\n\n**Transition Band**\nControls the width of the transition band of the bandpass filters used to split the signal in several bands. Higher values will use less CPU. But will result in more overlap between the bands.\n\n## References\n\n- [Wikipedia Dynamic Range](https://en.wikipedia.org/wiki/Dynamic_range)\n- [Crest factor](https://en.wikipedia.org/wiki/Crest_factor)\n- [Kurtosis](https://en.wikipedia.org/wiki/Kurtosis)\n- [Spectral flux](https://en.wikipedia.org/wiki/Spectral_flux)\n","isInternal":false,"tokens":436,"sizeBytes":2133},{"name":"deepfilternet.md","path":"src/contents/docs/plugins/deepfilternet.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/deepfilternet.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Deep Noise Remover\n\nAdvanced noise reduction is achieved using \"DeepFilterNet\" technology. This technology uses AI technology called deep learning to efficiently remove noise from audio signals, enabling higher quality noise suppression than conventional methods. This technology has several control parameters, which are explained below and how to set them.\n\n**Model** \nSelects the noise reduction model. Each one must be installed separately.\n\n- DeepFilterNet: requires [DeepFilterNet](https://github.com/Rikorose/DeepFilterNet).\n- DPDFNet: requires [HushMic](https://github.com/Fovty/HushMic). The Attenuation Limit and Process as Mono controls apply to this model.\n\n**Process as Mono (DPDFNet only)**\n\nAverages the left and right input channels before resampling, processes the result with one model instance, and sends the same processed signal to both output channels. This can significantly reduce CPU usage for mono microphones that supply the same signal on both channels. It does not change the selected model or its noise reduction quality.\n\nThis setting is off by default and is saved in presets. Enable it for mono speech input; leave it off when stereo separation matters. If only one input channel contains audio, averaging reduces its level by 6 dB. Opposite-polarity signals can cancel, so check the input channels before enabling it.\n\n**Attenuation Limit** \nThis parameter determines how much noise is attenuated. Higher values result in stronger noise reduction, but too high a value may also remove parts of the audio.\n\n- Recommended setting: 70dB to 80dB is a typical balanced value. This minimizes background noise while preserving speech intelligibility.\n\n**Minimum Processing Threshold** \nSignals below the Minimum Processing Threshold are targeted for denoising. Signals louder than this threshold are not subject to noise reduction and are passed through unchanged. This setting is intended to effectively remove small background noise.\n\n- Recommended setting: -15dB to -30dB is good. This should be fine-tuned according to the ambient noise. Currently, Easy Effects does not allow a value smaller than -15dB.\n\n**Maximum ERB Processing Threshold** \nThe human auditory system has the ability to separate and recognize different frequency components. Equivalent Rectangular Bandwidth (ERB) is a numerical expression and model of this filtering capability. ERB-based processing emphasizes the perceptual importance of speech by focusing on specific frequency bands. This improves the effectiveness of noise reduction and speech enhancement. This threshold sets the maximum level at which perceptually equivalent bandwidth (ERB)-based processing is applied. Signals above this threshold are not subject to noise reduction or speech enhancement processing. This prevents important speech components or loud portions of speech from being unnecessarily processed.\n\n- Recommended setting: Start in the range of 20dB to 30dB and adjust as needed.\n\n**Maximum DF Processing Threshold** \nThe Maximum DF Processing Threshold sets the maximum level of signal that DeepFilterNet will process. Signals exceeding this threshold are excluded from noise reduction processing. This prevents important or loud portions of speech from being unnecessarily processed.\n\n- Recommended setting: 20 dB as the default value. This effectively suppresses noise while maintaining voice quality.\n\n**Minimum Processing Buffer** \nDefines the size of the buffer used to process frames of audio data. A larger buffer can improve noise suppression but may increase latency.\n\n- Recommended setting: Set to a small value (0) if real-time processing is important, or to a number greater than 1 if noise suppression accuracy is a priority.\n\n**Post Filter Beta** \nControls the intensity of the post-processing filter applied after the initial noise suppression. This allows for more subtle refinement of the audio signal.\n\n- Recommended setting: 0.5dB to 2dB is generally recommended, but currently Easy Effects has a maximum value of 0.05dB, which is almost ineffective.\n","isInternal":false,"tokens":780,"sizeBytes":4064},{"name":"deesser.md","path":"src/contents/docs/plugins/deesser.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/deesser.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Deesser\n\nA Deesser is used to dynamically reduce high frequencies. The standard field of use of this plugin is the reduction of \"sssss\" and \"shhhh\" in vocal tracks. Easy Effects uses the Deesser developed by Calf Studio Gear.\n\n**Detection** \nSelect the detection of the Sidechain signal between Peak (stronger) and RMS (smoother).\n\n**Mode** \nSelect the operation mode between Wideband and Split. In Split mode not the full range signal will be affected by the gain reduction, but only frequencies above the split frequency will be manipulated in gain.\n\n**F1 Split** \nThe split frequency. All signals above this frequency will affect the gain reduction (and are affected in Split mode too).\n\n**F1 Gain** \nIt shifts the volume of the higher band. In Wideband mode it affects the Sidechain. In Split mode it also affects the processed high frequencies.\n\n**F2 Peak** \nCenter frequency of the bell filter. It allows a more precise selection of the Sidechain signal.\n\n**F2 Level** \nIncreases or decreases the level of the chosen F2 frequency.\n\n**F2 Peak Q** \nSet the quality of the bell filter. Higher values will affect a narrower frequency range. Lower values will affect a wider band.\n\n**Laxity** \nThe reaction of the Deesser. Higher values won't affect really short peaks.\n\n**Threshold** \nThe level above which the gain reduction is applied.\n\n**Ratio** \nThe amount of attenuation applied to the signal.\n\nFor example, a Ratio of 2 means that if the level rises 4 dB above the Threshold, it will be only 2 dB above after the reduction.\n\n**Makeup** \nThe gain to apply after the processing stage. In Split mode only the high band will be made up.\n\n## References\n\n- [Wikipedia De-essing](https://en.wikipedia.org/wiki/De-essing)\n- [Calf Deesser](https://calf-studio-gear.org/doc/Deesser.html)\n- [LedgerNote - De-esser: The Guide for Sibilant-Free Vocal Recordings](https://ledgernote.com/columns/mixing-mastering/de-esser/)\n","isInternal":false,"tokens":453,"sizeBytes":1930},{"name":"delay.md","path":"src/contents/docs/plugins/delay.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/delay.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Delay\n\nThis plugin allows the user to add a short Delay of time to each individual channel of the stereo stream. Easy Effects Delay uses the Delay Compensator Stereo by Linux Studio Plugins set in time mode.\n\n**Left Delay** \nLeft channel delay in milliseconds.\n\n**Right Delay** \nRight channel delay in milliseconds.\n\n**Left Dry Level** \nAmount of right unprocessed signal mixed in the output.\n\n**Right Dry Level** \nAmount of right unprocessed signal mixed in the output.\n\n**Left Wet Level** \nAmount of left processed signal mixed in the output.\n\n**Right Wet Level** \nAmount of right processed signal mixed in the output.\n\n## References\n\n- [Wikipedia Delay (audio effect)](https://en.wikipedia.org/wiki/Delay_(audio_effect))\n- [LSP Delay Compensator Stereo](http://lsp-plug.in/?page=manuals§ion=comp_delay_x2_stereo)\n","isInternal":false,"tokens":188,"sizeBytes":828},{"name":"echocanceller.md","path":"src/contents/docs/plugins/echocanceller.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/echocanceller.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Echo Canceller\n\nThe Echo is a reflected sound wave with sufficient magnitude and delay to be detectable as a signal distinct from the source one. An Echo Canceller is used to improve voice quality by preventing Echo from being created or removing it after it has been added to the source signal. Easy Effects uses the Echo Canceller from SpeexDSP library.\n\n**Frame Size** \nThe amount of time in milliseconds to process at once. It is recommended to use a frame size in the order of 20 ms.\n\n**Filter Length** \nThe amount of time of the Echo cancelling filter to use (also known as tail length). The recommended tail length is approximately the third of the room reverberation time. For example, in a small room, reverberation time is in the order of 300 ms, so a tail length of 100 ms is a good choice.\n\n## References\n\n- [Wikipedia Echo Suppression and Cancellation](https://en.wikipedia.org/wiki/Echo_suppression_and_cancellation)\n- [Speex Acoustic Echo Canceller](https://www.speex.org/docs/manual/speex-manual/node4.html#SECTION00450000000000000000)\n","isInternal":false,"tokens":242,"sizeBytes":1056},{"name":"equalizer.md","path":"src/contents/docs/plugins/equalizer.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/equalizer.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Equalizer\n\nThe Equalization in sound recording and reproduction is the process of adjusting the volume of different frequency bands within an audio signal. Easy Effects uses the Parametric Equalizer from Linux Studio Plugins. The user can choose from 1 to 32 bands. Width and center frequency of each band can be customized as needed.\n\n## Global Options\n\n**Bands** \nThe number of bands.\n\n**Mode** \n- **IIR** - Infinite Impulse Response filters, nonlinear minimal phase. In most cases does not add noticeable latency to output signal.\n- **FIR** - Finite Impulse Response filters with linear phase, finite approximation of Equalizer's impulse response. Adds noticeable latency to output signal.\n- **FFT** - Fast Fourier Transform approximation of the frequency chart, linear phase. Adds noticeable latency to output signal.\n\n**Balance** \nBalance between left and right output channels.\n\n**Pitch Left** \nThe frequency shift for all filters of the left channel, in semitones.\n\n**Pitch Right** \nThe frequency shift for all filters of the right channel, in semitones.\n\n**Split Channels** \nWhen enabled it is possible to apply different configurations to left and right channels.\n\n**Flat Response** \nThis function sets each band gain to 0.\n\n**Calculate Frequencies** \nThis function calculates the center frequency and the width of each band using the current number of bands. Useful when the user wants fewer than 32 bands but has no idea about which frequencies should be chosen.\n\n## Band Options\n\n**Type** \n- **Off** - The filter is not working (turned off).\n- **Bell** - Bell filter with smooth peak/recess.\n- **High Pass** - High Pass filter with rejection of low frequencies.\n- **High Shelf** - Shelving filter with adjustment of high frequency range.\n- **Low Pass** - Low Pass filter with rejection of high frequencies.\n- **Low Shelf** - Shelving filter with adjustment of low frequency range.\n- **Notch** - Notch filter with full rejection of selected frequency.\n- **Resonance** - Resonance filter with sharp peak/recess.\n- **All Pass** - All Pass filter.\n\n**Mode** \n- **RLC** - Very smooth filters based on similar cascades of RLC contours. Bilinear Z-transform (BT) or Matched Z-transform (MT) is used for pole/zero mapping.\n- **BWC** - Butterworth-Chebyshev-type-1 based filters. Does not affect Resonance and Notch filters. Bilinear Z-transform (BT) or Matched Z-transform (MT) is used for pole/zero mapping.\n- **LRX** - Linkwitz-Riley based filters. Does not affect Resonance and Notch filters. Bilinear Z-transform (BT) or Matched Z-transform (MT) is used for pole/zero mapping.\n- **APO** - Digital biquad filters derived from canonic analog biquad prototypes digitalized through Bilinear transform. These are [textbook filters](https://shepazu.github.io/Audio-EQ-Cookbook/audio-eq-cookbook.html) which are implemented as in the [Equalizer APO](https://equalizerapo.com/) software. Direct design (DR) is used to serve the digital filter coefficients directly in the digital domain, without performing transforms.\n\n**Slope** \nThe slope of the filter characteristics.\n\n**Solo** \nMakes the selected band the only one active.\n\n**Mute** \nMutes the selected band.\n\n**Frequency** \nCenter frequency of the selected band.\n\n**Width** \nBandwidth calculated as `width = frequency / quality`.\n\n**Quality** \nThe quality factor of the filter used.\n\n## References\n\n- [Wikipedia Equalization (audio)](https://en.wikipedia.org/wiki/Equalization_(audio))\n- [LSP Parametric Equalizer x32 LeftRight](http://lsp-plug.in/?page=manuals§ion=para_equalizer_x32_lr)\n- [Wikipedia Q Factor](https://en.wikipedia.org/wiki/Q_factor)\n- [How to EQ - Q Factor and Bandwidth in EQ: What They Mean](https://howtoeq.wordpress.com/2010/10/07/q-factor-and-bandwidth-in-eq-what-it-all-means/)\n","isInternal":false,"tokens":885,"sizeBytes":3781},{"name":"exciter.md","path":"src/contents/docs/plugins/exciter.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/exciter.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Exciter\n\nAn Exciter is used to produce high sound that is not present in the original signal. This is done by creating harmonic distortions of the signal which are restricted in range and added to the original signal. It raises the upper end of an audio signal without simply raising the higher frequencies like an [Equalizer](equalizer.md) would do to create a more \"crisp\" or \"brilliant\" sound.\n\n**Amount** \nAmount of harmonics added to the original signal.\n\n**Harmonics** \nAmount of newly created harmonics.\n\n**Scope** \nThe frequency above which harmonics are produced.\n\n**Ceiling (button)** \nConstrain the excitement on the upper end.\n\n**Ceiling (value)** \nThe frequency above which no harmonics are produced.\n\n**Blend Harmonics** \nThe \"colour\" (or octave) of the harmonics.\n\n**Listen** \nMute the original signal and listen to the harmonics exclusively.\n\n## References\n\n- [Calf Exciter](https://calf-studio-gear.org/doc/Exciter.html)\n- [Wikipedia Exciter (effect)](https://en.wikipedia.org/wiki/Exciter_(effect))\n","isInternal":false,"tokens":244,"sizeBytes":1026},{"name":"expander.md","path":"src/contents/docs/plugins/expander.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/expander.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Expander\n\nAn expander is an effect that works inversely to a compressor, further reducing the volume of quiet sounds below a certain level. This allows for natural suppression of background noise and unwanted whispers. For example, it is effective when air conditioning or other environmental sounds intrude into quiet scenes.\n\nThis effect is similar to the gate, which is explained in another chapter, in that it blocks quiet sounds. However, while a gate completely cuts sound, an expander gradually reduces sound, resulting in a more natural and smooth decay characteristic. Its main uses are listed below:\n\n- Reduce the prominence of quiet environmental noises (air conditioning, PC fans, clicks, etc.) during periods of silence\n- Suppress noise in quiet sections while maintaining the clarity of narration and dialogue\n- Emphasize the contours (punch) of sounds to create a more dynamic impression\n\n## Expander Options\n\nThe following is an explanation of each parameter when the mode is set to \"**Downward**.\"\n\n**Attack Time**: This determines the time it takes for the expander to begin suppressing the sound after the sound volume drops below the threshold. Shorter values will reduce the volume immediately, quickly suppressing noise and other imperfections. Longer values will cause the processing to begin later, preserving the natural reverberation and nuances of soft sounds.\n\n**Release Time**: This determines the time it takes for the expander to stop suppressing the volume and return to its original volume after the sound exceeds the threshold again and is determined to no longer require processing.\n\nLonger values will cause the sound to return more slowly, making it sound more natural.\n\nShorter values will return the sound to its original volume quickly, making it sound clearer and tighter, but may sound unnatural in some cases.\n\n**Attack Threshold**: The effect is only applied to sounds quieter than this value.\n\n**Release Threshold**: This can be the same as the Attack Threshold, but setting it about -5dB lower than the Attack Threshold will enable the Hysteresis function, suppressing fluttering around the threshold and achieving more stable processing.\n\n**Ratio**: This is an indicator of how much quieter (= how much) sounds below the threshold will be reduced. Set the value using the following as a guide:\n\n- A setting of 1 passes the signal as is (same as Expander OFF)\n- For light noise suppression (natural processing), use 1.5-3\n- For clarity of dialogue and smoothing out quiet sections, use 4-8\n- For complete noise reduction (gate-like behavior), use 20-100\n\n*Please note that setting the ratio too high can result in a choppy, unnatural sound.\n\n**Knee**: This parameter adjusts how smoothly processing begins near the threshold. The name comes from the fact that the curve on the graph resembles a knee. See below for the relationship between the setting value and the curve.\n\nKnee Value Volume Curve Characteristics Processing Sensation\n\n0dB Abrupt bend (hard knee) The effect is applied suddenly (like an ON/OFF switch)\n\n-6dB Gradual bend Processing begins naturally\n\n-12dB and above Very smooth Processing is barely noticeable\n\n**Makeup**: The gain applied after gating.\n\n**Dry Level**: The amount of unprocessed signal mixed with the output.\n\n**Wet Level**: The amount of processed signal mixed with the output.\n\nFor information on how to set up sidechaining and sidechain filtering, see the chapter of Gate.\n\n*When \"**Upward**\" is selected in the mode setting, the expander will function to boost (increase the volume of) quieter sounds.\n\nThis can be used in situations where dialogue or narration is too quiet to be heard, and is also effective when you want to bring out the subtle nuances of a musical performance (such as pianissimo).\n\nRather than \"lowering louder sounds\" like a compressor, it is used to increase the average volume by \"boosting quieter sounds,\" or to increase sound pressure while maintaining the punch and three-dimensionality of the original sound.","isInternal":false,"tokens":811,"sizeBytes":4023},{"name":"filter.md","path":"src/contents/docs/plugins/filter.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/filter.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Filter\n\nA Filter is used to amplify (boost), pass or attenuate (cut) defined parts of a frequency spectrum. Easy Effects uses the Filter from Linux Studio Plugins.\n\n**Type** \n- **Low-pass** - Low-pass filter with rejection of high frequencies. \n- **High-pass** - High-pass filter with rejection of low frequencies. \n- **Low-shelf** - Shelving filter with adjustment of low frequencies. \n- **High-shelf** - Shelving filter with adjustment of high frequency range. \n- **Bell** - Bell filter with smooth peak/recess. \n- **Bandpass** - Bandpass filter. \n- **Notch** - Notch filter with full rejection of selected frequency. \n- **Resonance** - Resonance filter with sharp peak/recess. \n- **Ladder-pass** - The filter that makes some ladder-passing in the spectrum domain. \n- **Ladder-rej** - The filter that makes some ladder-rejection in the spectrum domain. \n- **Allpass** - All-pass filter which only affects the phase of the audio signal at the specified frequency. \n\n**Filter Mode** \n- **RLC** - Very smooth filters based on similar cascades of RLC contours. \n- **BWC** - Butterworth-Chebyshev-type-1 based filters. Does not affect Resonance and Notch filters. \n- **LRX** - Linkwitz-Riley based filters. Does not affect Resonance and Notch filters. \n- **APO** - Digital biquad filters derived from canonic analog biquad prototypes digitalized through Bilinear transform. These are textbook filters which are implemented as in the EqualizerAPO software. \n- **BT** - Bilinear Z-transform is used for pole/zero mapping. \n- **MT** - Matched Z-transform is used for pole/zero mapping. \n- **DR** - Direct design is used to serve the digital filter coefficients directly in the digital domain, without performing transforms. \n\n**Equalizer Mode** \n- **IIR** - Infinite Impulse Response filters, nonlinear minimal phase. In most cases does not add noticeable latency to output signal. \n- **FIR** - Finite Impulse Response filters with linear phase, finite approximation of equalizer's impulse response. Adds noticeable latency to output signal. \n- **FFT** - Fast Fourier Transform approximation of the frequency chart, linear phase. Adds noticeable latency to output signal. \n- **SPM** - Spectral Processor Mode of equalizer, equalizer transforms the magnitude of signal spectrum instead of applying impulse response to the signal.\n\n**Slope** \nThe slope of the filter characteristics. \n\n**Frequency** \nThe cutoff/resonance frequency of the filter or the middle frequency of the band. \n\n**Width** \nThe width of the bandpass/ladder filters in octaves. \n\n**Gain** \nThe gain of the filter. It is disabled for lo-pass/hi-pass/notch filters. \n\n**Quality** \nThe quality factor of the filter. \n\n**Balance** \nThe balance between left and right output channels. \n\n## References\n\n- [Wikipedia Audio Filter](https://en.wikipedia.org/wiki/Audio_filter)\n- [Linux Studio Plugins Filter](https://lsp-plug.in/?page=manuals§ion=filter_stereo)\n","isInternal":false,"tokens":692,"sizeBytes":2958},{"name":"gate.md","path":"src/contents/docs/plugins/gate.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/gate.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Gate\n\nThe Gate attenuates signals that register below a Threshold. This kind of signal processing is used to reduce disturbing noise between useful signals. Easy Effects uses the Stereo Sidechain Gate from Linux Studio Plugins.\n\n## Gate Workflow\n\nThe Gate begins to open when the Sidechain level becomes above the Attack Zone Start level.\n\nThe Gate fully opens when the Sidechain level becomes above the Attack Threshold level.\n\nThe Gate begins to close when the Sidechain level becomes below the Release Zone Start level.\n\nThe Gate fully closes when the Sidechain level becomes below the Release Threshold level.\n\n## Gate Options\n\n**Attack Time** \nThe length of time it takes to restore roughly two-thirds of the gain reduction.\n\n**Release Time** \nThe length of time it takes to apply roughly two-thirds of the gain reduction.\n\n**Curve Threshold** \nThe Gate fully opens upon the Sidechain level becoming above Curve Threshold (displayed as Attack Threshold level).\n\nIf Hysteresis is not enabled, the Gate begins to close upon the Sidechain level becoming below Curve Threshold (displayed as Release Zone Start level).\n\n**Curve Zone Size** \nThe Gate begins to open upon the Sidechain level becoming above the Curve Threshold + Curve Zone (displayed as Attack Zone Start level).\n\nIf Hysteresis is not enabled, the Gate fully closes upon the Sidechain level becoming below the Curve Threshold + Curve Zone (displayed as Release Threshold level).\n\n**Hysteresis** \nWhen enabled, Curve Threshold and Curve Zone apply only to the opening Gate, and separate parameters can be configured for closing Gate.\n\n**Hysteresis Threshold** \nIf Hysteresis is enabled, the Gate begins to close upon the Sidechain level becoming below Curve Threshold + Hysteresis Threshold (displayed as Release Zone Start level).\n\n**Hysteresis Zone Size** \nIf Hysteresis is enabled, the Gate fully closes upon the Sidechain level becoming below the Curve Threshold + Hysteresis Threshold + Hysteresis Zone (displayed as Release Threshold level).\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal mixed in the output.\n\n**Reduction** \nIf the value is negative, it acts as the amount of gain reduction to apply to the input signal when the Gate is fully closed. If the value is positive, the Gate operates in \"Reverse Mode\": It reduces the amplitude when the signal is above the threshold.\n\n**Makeup** \nThe gain to apply after the gating stage.\n\n## Sidechain\n\n**Listen** \nAllows to listen the processed Sidechain signal.\n\n**Input Type** \nDetermines which signal is the Sidechain or, in other words, the signal that controls the gating stage.\n\n- **Internal** - The Sidechain is the Gate input signal (taken after applying the plugin input gain).\n- **External** - The Sidechain is an external source took by a specific input device (typically a microphone).\n\n**Input Device** \nSelect the device for the External Sidechain.\n\n**Mode** \nDetermines how the Sidechain is evaluated for the gating stage.\n\n- **Peak** - The Gate reacts according to the peaks.\n- **RMS** - The Gate reacts according to the average loudness measured by the root mean square.\n- **Low-Pass Filter** - The Gate reacts according to the signal processed by a recursive 1-pole Low-Pass filter.\n- **Simple Moving Average** - The Gate reacts according to the signal processed by the Simple Moving Average filter.\n\n**Source** \nDetermines which part of the Sidechain is taken into account for the gating stage.\n\n- **Middle** - The sum of left and right channels.\n- **Side** - The difference between left and right channels.\n- **Left** - Only left channel is used.\n- **Right** - Only right channel is used.\n- **Min** - The absolute minimum value is taken from stereo input.\n- **Max** - The absolute maximum value is taken from stereo input.\n\n**PreAmplification** \nGain applied to the Sidechain signal.\n\n**Reactivity** \nThe time that defines the number of samples used to process the Sidechain in RMS, Uniform and Low-Pass modes. Higher the value, more smooth the gating.\n\n**Lookahead** \nThe signal to gate is delayed by this amount of time, so that the gating will be applied earlier than it would be otherwise. The corresponding delay is reproduced on the output signal.\n\n## Sidechain Filters\n\n**High-Pass Filter Mode** \nSets the type of the High-Pass filter applied to Sidechain signal.\n\n**High-Pass Frequency** \nSets the cut-off frequency of the High-Pass filter.\n\n**Low-Pass Filter Mode** \nSets the type of the Low-Pass filter applied to Sidechain signal.\n\n**Low-Pass Frequency** \nSets the cut-off frequency of the Low-Pass filter.\n\n## References\n\n- [Wikipedia Dynamic Range Compression](https://en.wikipedia.org/wiki/Dynamic_range_compression)\n- [LSP Sidechain Gate Stereo](https://lsp-plug.in/?page=manuals§ion=sc_gate_stereo)\n- [Wikipedia Noise Gate](https://en.wikipedia.org/wiki/Noise_gate)\n","isInternal":false,"tokens":1070,"sizeBytes":4911},{"name":"harmonizer.md","path":"src/contents/docs/plugins/harmonizer.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/harmonizer.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Harmonizer\n\nThe Harmonizer adds two pitch shifted voices to the input. Set the root note and chord, then choose an interval and level for each voice. The dry/wet control balances the original signal with the harmonized signal.\n\nEasy Effects uses the StereoHarm LV2 plugin from Rakarrack Plus. Install Rakarrack Plus to use this effect.\n\n## References\n\n- [Rakarrack Plus](https://github.com/Stazed/rakarrack-plus)\n- [Harmonizer documentation](https://github.com/Stazed/rakarrack-plus)\n","isInternal":false,"tokens":116,"sizeBytes":486},{"name":"limiter.md","path":"src/contents/docs/plugins/limiter.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/limiter.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Limiter\n\nA Limiter is a special type of downward [Compressor](compressor.md) which does not allow the signal to overtake a predetermined Threshold. Ideally it has a very high compression ratio that takes the amplitude below a ceiling which stands as the maximum output level. For this reason it is usually named \"brick-wall limiter\".\n\nEasy Effects uses the Sidechain Stereo Limiter from Linux Studio Plugins. In most cases it works as a brick-wall limiter, but it offers also an additional feature that acts like a Compressor with extreme settings, so the output signal may exceed the specified Threshold.\n\n**Mode** \nSelect the operative mode of the peak cutting algorithm which searches the peaks above the Threshold and applies short gain reduction patches to the signal.\n\nThese patches can be selected in 3 forms: **Hermite**, **Exponential** and **Linear**. Each one has 4 different variants related to gain reduction of the samples around the peak: **Thin**, **Tail**, **Duck** and **Wide**.\n\nThe shape of each form and variant can be referred into the Linux Studio Plugin manual. See References section at the bottom of the present document.\n\n**Oversampling** \nWhen enabled, the sample rate is internally increased in order to improve peak detection and reduce aliasing (that causes distortion).\n\nThe modes have 2 main types: **Full**, which increases both the Sidechain and the Input signals, and **Half** which increases only the Sidechain. Each one of them specifies different **multipliers** and, between parentheses, the number of \"**lobes** in the kernel\".\n\nThe oversampled Input signal is downsampled to the original sample rate after processing.\n\n**Dither** \nIf specified, it enables Dithering for the selected bit depth. The process of Dithering adds a low-level noise to output signal in order to mask \"quantization distortion\", a form of artifact present in digital audio rendered at lower bit depth.\n\n**SC PreAmp** \nThe gain applied to the Sidechain before it is processed.\n\n**Lookahead** \nThe size of the buffer used to detect the peaks in advance. It adds the corresponding latency to the output signal.\n\n**Attack** \nThe length of time it takes to apply the needed gain reduction to keep the peak below the Threshold.\n\nIt affects the length of the gain reduction patch. It cannot be larger than the Lookahead (if specified larger, it's set internally as the max possible value).\n\n**Release** \nThe length of time it takes to restore the reduced gain around the limited peak.\n\nIt affects the length of the gain reduction patch. It cannot be twice larger than the Lookahead (if specified larger, it's set internally as the max possible value).\n\n**Threshold** \nThe target level above which the Limiter should reduce the peak of the signal. In some modes it represents the maximum output level.\n\n**Threshold Boost** \nIf enabled it applies an amount of gain to the limited signal equal to the absolute value of the Threshold. This causes the peak limited at the Threshold level to output at 0 dB (it has only effect when the Threshold is set below 0 dB).\n\n**Stereo Link** \nThe degree of the channel linking. At 0% both channels are limited independently while at 100% the loudest one triggers the same gain reduction on both.\n\n**External Sidechain** \nSwitch the Sidechain to an external source took by a specific input device (typically a microphone).\n\n## Auto Leveling\n\nThe Auto Leveling checkbutton introduces an additional feature named \"Automated Level Regulation\" (ALR) which acts like a Compressor with infinite ratio for the purpose of applying a smoothed gain reduction rather than a stronger peak cutter like in brick-wall mode. This configuration could get the output level to exceed the Threshold even if the signal is highly compressed.\n\n**Attack** \nManage how the raise of the input signal affects the smoothness of the ALR curve that controls the gain reduction level. Higher the value, more quickly the curve goes to it's maximum.\n\n**Release** \nManage how the fall of the input signal affects the smoothness of the curve that controls the gain reduction level. Higher the value, more quickly the curve goes to it's minimum.\n\n**Knee** \nManage the Threshold of the ALR gain curve and, in fact, adjust the balance between two gain reduction stages. Raising the value delegates more work to the peak-cutting algorithm. Lowering the value delegates more work to the ALR gain reduction algorithm.\n\n## References\n\n- [Wikipedia Limiter](https://en.wikipedia.org/wiki/Limiter)\n- [LSP Sidechain Limiter Stereo](https://lsp-plug.in/?page=manuals§ion=sc_limiter_stereo)\n","isInternal":false,"tokens":971,"sizeBytes":4607},{"name":"loudness.md","path":"src/contents/docs/plugins/loudness.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/loudness.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Loudness\n\nEasy Effects uses the Loudness Compensator from Linux Studio Plugins which applies the \"equal-loudness contour\" corrections to the input signal.\n\nAn equal-loudness contour is a measure of sound pressure level (SPL), over the frequency spectrum, for which a listener perceives a constant loudness when presented with pure steady tones. The unit of measurement for loudness levels is the phon and is arrived at by reference to equal-loudness contours.\n\nUsage of equal-loudness contours solves many mixing problems that every sound engineer meets while working on the track. The main problem is that human ear perceives different frequencies for different volume settings in a different way. In other words, applying changes to the mix on the low volume settings may cause unexpected sounding of the mix at the maximum loudness.\n\nThe Loudness Compensator performs frequency response computations and applies the computed frequency response to the input signal depending on the output volume settings. Additionally it can provide ear protection by applying hard-clipping to the output signal if it exceeds the allowed configurable level.\n\n**Standard** \nAllows to select different equal-loudness contour standards.\n\n- **Flat** - Applies flat frequency response to the whole spectrum. It's similar to just a gain knob but useful to perform a comparison to other modes.\n- **ISO 226:2003** - Recent equal-loudness contour standard published in 2003.\n- **Fletcher-Munson** - The first equal-loudness contour implementation by Harvey Fletcher and Wilden A. Munson published in 1933.\n- **Robinson-Dadson** - More accurate equal-loudness contour implementation by D.W. Robinson and R.S. Dadson published in 1956. It became the basis for the ISO 226:2003 standard.\n\n**FFT Size** \nAllows to select size of the Fast Fourier Transform frame used for the processing. The larger FFT frame is, the more precise the curve is approximated and the more latency is introduced.\n\n**Output Volume** \nThe output volume of the signal with applied equal loudness contour. It controls the loudness of the 1 kHz pure sine wave.\n\n**Clipping** \nAllows to enable and the hard clipping of the output signal.\n\n**Clipping Range** \nAllows to set the gap level for the hard clipping of the output signal.\n\n## References\n\n- [Wikipedia Equal-Loudness Contour](https://en.wikipedia.org/wiki/Equal-loudness_contour)\n- [LSP Loudness Compensator Stereo](https://lsp-plug.in/?page=manuals§ion=loud_comp_stereo)\n- [Lindos Electronics - Equal-Loudness Contours](http://www.lindos.co.uk/cgi-bin/FlexiData.cgi?SOURCE=Articles&VIEW=full&id=17)\n","isInternal":false,"tokens":577,"sizeBytes":2614},{"name":"maximizer.md","path":"src/contents/docs/plugins/maximizer.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/maximizer.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Maximizer\n\nThe Maximizer is a special type of [Limiter](limiter.md) that does not only prevent the signal to exceed a specified target level, but also adjusts the average Loudness of the audio track.\n\nEasy Effects uses the Maximizer developed by ZamAudio. It acts like an Amplifier that feeds a brick-wall Limiter with a fixed Lookahead of 10 ms adding the corresponding delay to the output signal.\n\n**Threshold** \nThis parameter represents the ideal amplification level needed by the signal.\n\nIt contributes along with the Ceiling to determine the amount of gain to apply to the signal before the limiting stage. The gain is calculated by `(-Threshold) - (-Ceiling)`.\n\nFor example, on Threshold -6 dB and Ceiling -2 dB, the signal is amplified by 4 dB and limited to not exceed -2 dB output level.\n\n**Ceiling** \nThis parameter represents the ideal attenuation level needed by the signal and the maximum allowed output level.\n\nWhen the Threshold is set to 0 dB, the Ceiling is simply the gain reduction. When the Threshold is lowered, the signal is boosted without overtaking the Ceiling value.\n\n**Release** \nSets the release of the internal brick-wall Limiter. Lower values may introduce small artifacts.\n\n## References\n\n- [Sonarworks Blog - What Is A Maximizer?](https://www.sonarworks.com/soundid-reference/blog/learn/what-is-a-maximizer/)\n","isInternal":false,"tokens":293,"sizeBytes":1348},{"name":"midside_equalizer.md","path":"src/contents/docs/plugins/midside_equalizer.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/midside_equalizer.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Mid-Side Equalizer\n\nThe Mid-Side Equalizer processes the Mid (sum) and Side (difference) parts of a stereo signal independently. It's useful, for example, to fix the overall tone of a recording through the Mid channel while keeping the stereo ambience untouched on the Side channel. Easy Effects uses the Parametric Equalizer x32 MidSide from Linux Studio Plugins. You can run anywhere from 1 to 32 bands per channel, and tweak the width and center frequency of each band.\n\n## Global Options\n\n**Bands** \nNumber of bands.\n\n**Mode** \n- **IIR** - Infinite Impulse Response filters, nonlinear minimal phase. Usually doesn't add noticeable latency to the output.\n- **FIR** - Finite Impulse Response filters with linear phase, a finite approximation of the equalizer's impulse response. Adds noticeable latency to the output.\n- **FFT** - Fast Fourier Transform approximation of the frequency chart, linear phase. Adds noticeable latency to the output.\n\n**Balance** \nBalance between the left and right output channels.\n\n**Pitch Mid** \nFrequency shift for all mid channel filters, in semitones.\n\n**Pitch Side** \nFrequency shift for all side channel filters, in semitones.\n\n**Link Mid/Side** \nWhen enabled, you can apply different configurations to the mid and side channels.\n\n**Flat Response** \nResets all band gains to 0.\n\n**Calculate Frequencies** \nFigures out the center frequency and width for each band based on the current band count. Handy if you want fewer than 32 bands but aren't sure which frequencies to pick.\n\n## Band Options\n\n**Type** \n- **Off** - Filter is disabled.\n- **Bell** - Bell filter with smooth peak/recess.\n- **High Pass** - High Pass filter that removes low frequencies.\n- **High Shelf** - Shelving filter that adjusts the high frequency range.\n- **Low Pass** - Low Pass filter that removes high frequencies.\n- **Low Shelf** - Shelving filter that adjusts the low frequency range.\n- **Notch** - Notch filter that fully rejects the selected frequency.\n- **Resonance** - Resonance filter with sharp peak/recess.\n- **All Pass** - All Pass filter.\n\n**Mode** \n- **RLC** - Very smooth filters built from cascaded RLC contours. Uses Bilinear Z-transform (BT) or Matched Z-transform (MT) for pole/zero mapping.\n- **BWC** - Butterworth-Chebyshev-type-1 based filters. Doesn't affect Resonance and Notch filters. Uses Bilinear Z-transform (BT) or Matched Z-transform (MT) for pole/zero mapping.\n- **LRX** - Linkwitz-Riley based filters. Doesn't affect Resonance and Notch filters. Uses Bilinear Z-transform (BT) or Matched Z-transform (MT) for pole/zero mapping.\n- **APO** - Digital biquad filters derived from canonical analog biquad prototypes digitalized through Bilinear transform. These are [textbook filters](https://shepazu.github.io/Audio-EQ-Cookbook/audio-eq-cookbook.html) implemented the same way as in [Equalizer APO](https://equalizerapo.com/). Direct design (DR) generates the digital filter coefficients directly in the digital domain, without any transforms.\n\n**Slope** \nSlope of the filter curve.\n\n**Solo** \nMakes the selected band the only active one.\n\n**Mute** \nMutes the selected band.\n\n**Frequency** \nCenter frequency of the band.\n\n**Width** \nBandwidth, calculated as `width = frequency / quality`.\n\n**Quality** \nQuality factor of the filter.\n\n## References\n\n- [Wikipedia Equalization (audio)](https://en.wikipedia.org/wiki/Equalization_(audio))\n- [LSP Parametric Equalizer x32 MidSide](http://lsp-plug.in/?page=manuals§ion=para_equalizer_x32_ms)\n- [Wikipedia Q Factor](https://en.wikipedia.org/wiki/Q_factor)\n- [How to EQ - Q Factor and Bandwidth in EQ: What They Mean](https://howtoeq.wordpress.com/2010/10/07/q-factor-and-bandwidth-in-eq-what-it-all-means/)\n","isInternal":false,"tokens":895,"sizeBytes":3712},{"name":"multibandcompressor.md","path":"src/contents/docs/plugins/multibandcompressor.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/multibandcompressor.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Multiband Compressor\n\nEasy Effects uses the Sidechain Multiband Compressor Stereo developed by Linux Studio Plugins. Please refer to the [Compressor](compressor.md) documentation to comprehend the basic functionality of the dynamic range compression.\n\n## Global Options\n\n**Band Management** \nThis Compressor allows to split the input signal up to 8 bands. Each band is not attached to its strict frequency and can be controlled by completely different frequency range that can be obtained by applying Low-Cut and Hi-Cut filters to the Sidechain signal.\n\nThe first band is always enabled while the others can be activated if needed. When only the first band is enabled, the functionality is similar to the Singleband Sidechain Compressor. Otherwise the signal is split in more bands and each band is compressed individually. After the compression stage, the bands are mixed together to form the output result.\n\n**Operating Mode** \nDetermines how the input signal is split to obtain the different bands.\n\n- **Classic** - The original signal is split using crossover filters. After the compression stage, all bands become phase-compensated using All-Pass filters.\n- **Modern** - Each band is processed by a pair of dynamic shelving filters. This mode allows a better control over the gain of each band.\n\n**Sidechain Boost** \nIf enabled, it introduces a special mode for assigning the same weight for higher frequencies opposite to lower frequencies.\n\nWhen disabled, the frequency band is processed 'as is', but the usual audio signal has 3 dB/octave falloff in the frequency domain and could be compared with the pink noise. So lower frequencies take more effect on the Compressor rather than higher frequencies.\n\nOn the other hand the Sidechain Boost mode allows to compensate the -3 dB/octave falloff of the signal spectrum and, even more, makes the signal spectrum growing +3 dB/octave in the almost fully audible frequency range.\n\n**Pink** applies +3 dB/octave while **Brown** applies +6 dB/octave Sidechain Boost. Each of them can use bilinear-transformed (BT) or matched-transformed (MT) shelving filter.\n\n**Sidechain Source** \nIf the External Sidechain is enabled inside at least one band, this combobox allows to select the input device as source.\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal mixed in the output.\n\n## Band Options\n\n**Band Start** \nAllows to change the lower end split frequency of the selected band. This value is assigned to the Band End of the previous enabled band. It can be modified for all bands except the first one, which is always enabled and its value is 0 Hz.\n\n**Band End** \nSpecify the upper end split frequency of the selected band. It cannot be directly modified and assumes the same value of the Band Start of the next enabled band. For the last enabled band it is always 24.000 Hz.\n\n**Compression Mode** \n- **Downward** - It's aimed to decrease the gain of the signal above the Threshold.\n- **Upward** - It's aimed to increase the gain of the signal below the Threshold.\n- **Boosting** - It's aimed to increase the gain of the signal below the Threshold by a specific amount.\n\n**External Sidechain** \nThe Sidechain is an external source took by a specific input device (typically a microphone).\n\n**Band Bypass** \nIf enabled, the selected band is not affected by the compression stage.\n\n**Solo** \nTurns on the Solo mode to the selected band by applying -36 dB gain to the other non-soloing bands.\n\n**Mute** \nTurns on the Mute mode applying -36 dB gain to to the selected band.\n\n**Attack Time** \nThe length of time it takes to apply roughly two-thirds of the targeted amount of compression ratio to the uncompressed band signal.\n\n**Release Time** \nThe length of time it takes to restore roughly two-thirds of the reduced gain (in Downward mode) or increased gain (in Upward/Boosting mode) to the compressed band signal.\n\n**Attack Threshold** \nThe target level around which the compression is applied (the range depends by the Knee).\n\n**Release Threshold** \nSets up the Threshold of the Release Time, calculated by summing the Release Threshold to the Attack Threshold.\n\nIf the Sidechain level is above that Threshold, the compressor uses the Release Time for the releasing stage. Otherwise the Attack Time is used in place of Release Time.\n\nFor example, with -10 dB Attack Threshold and -60 dB Release Threshold, if the Sidechain is above `-10 + (-60) = -70 dB`, the Release Time is used for the gain restoration. If the Sidechain is below -70 dB, the Attack Time is used for the gain restoration.\n\n**Ratio** \nThe amount of attenuation (in Downward mode) or amplification (in Upward/Boosting mode) that will be applied to the signal.\n\nFor example, when the Ratio is 2 in Downward mode and the Sidechain rises above the Threshold by 10 dB, the signal would be ideally reduced by 5 dB (10/2 dB). In practice this behavior mostly depends on how the Multiband Compressor is designed and configured.\n\n**Knee** \nThe range over which the Compressor switches from no compression to almost the full ratio compression (the Threshold usually sits at the center of this transition zone).\n\n**Makeup** \nThe gain to apply after the compression stage.\n\n## Band Sidechain Options\n\n**Mode** \nDetermines how the Sidechain of the selected band is evaluated for the compression stage.\n\n- **Peak** - The Compressor reacts according to the peaks.\n- **RMS** - The Compressor reacts according to the average loudness measured by the root mean square.\n- **Low-Pass** - The Compressor reacts according to the signal processed by a Low-Pass filter.\n- **Uniform** - The Compressor reacts according to the loudness measured by the average of the absolute amplitude.\n\n**Source** \nDetermines which part of the Sidechain is taken into account for the compression stage.\n\n- **Middle** - The sum of left and right channels.\n- **Side** - The difference between left and right channels.\n- **Left** - Only left channel is used.\n- **Right** - Only right channel is used.\n- **Min** - The absolute minimum value is taken from stereo input.\n- **Max** - The absolute maximum value is taken from stereo input.\n\n**Low-Cut Filter** \nEnables a custom Low-Cut Filter for the selected band.\n\n**Low-Cut Frequency** \nSets the cut-off frequency of the custom Low-Cut filter. If it is disabled, the default Low-Cut filter assumes internally the Band Start frequency as the cut-off frequency.\n\n**Hight-Cut Filter** \nEnables a custom High-Cut Filter for the selected band.\n\n**Hight-Cut Frequency** \nSets the cut-off frequency of the custom High-Cut filter. If it is disabled, the default High-Cut filter assumes internally the Band End frequency as the cut-off frequency.\n\n**PreAmp** \nGain applied to the Sidechain signal of the selected band.\n\n**Reactivity** \nThe time that defines the number of samples used to process the Sidechain in RMS, Uniform and Low-Pass modes. Higher the value, more smooth the compression.\n\n**Lookahead** \nThe band signal to compress is delayed by this amount of time, so that the compression will be applied earlier than it would be otherwise.\n\nEach band can have different Lookahead values. To avoid phase distortions in the mixing stage, all the bands are automatically delayed for an individually calculated period of time.\n\n**Boost Amount** \nMaximum gain amplification to apply in Boosting Mode.\n\n**Boost Threshold** \nThe Threshold below which a constant amplification will be applied to the band signal in Upward Mode (it prevents from applying infinite amplification to very quiet signals).\n\n## References\n\n- [Wikipedia Dynamic Range Compression](https://en.wikipedia.org/wiki/Dynamic_range_compression)\n- [LSP Sidechain Multiband Compressor Stereo](https://lsp-plug.in/?page=manuals§ion=sc_mb_compressor_stereo)\n- [Black Ghost Audio - The Ultimate Guide to Compression](https://www.blackghostaudio.com/blog/the-ultimate-guide-to-compression)\n- [Attack Magazine - Demolishing The Myths of Compression](https://www.attackmagazine.com/features/columns/gregory-scott-demolishing-the-myths-of-compression/)\n","isInternal":false,"tokens":1761,"sizeBytes":8120},{"name":"multibandgate.md","path":"src/contents/docs/plugins/multibandgate.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/multibandgate.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Multiband Gate\n\nEasy Effects uses the Multiband Gate Stereo developed by Calf Studio Gear. Please refer to the [Gate](gate.md) documentation to comprehend the basic functionality of the gating process.\n\n## Global Options\n\n**Band Management** \nThis Gate allows to split the input signal up to 8 bands. Each band is not attached to its strict frequency and can be controlled by completely different frequency range that can be obtained by applying Low-Cut and Hi-Cut filters to the Sidechain signal.\n\nThe first band is always enabled while the others can be activated if needed. When only the first band is enabled, the functionality is similar to the Singleband Sidechain Gate. Otherwise the signal is split in more bands and each band is processed individually. After the gating stage, the bands are mixed together to form the output result.\n\n**Operating Mode** \nDetermines how the input signal is split to obtain the different bands.\n\n- **Classic** - The original signal is split using crossover filters. After the gating stage, all bands become phase-compensated using All-Pass filters.\n- **Modern** - Each band is processed by a pair of dynamic shelving filters. This mode allows a better control over the gain of each band.\n\n**Sidechain Boost** \nIf enabled, it introduces a special mode for assigning the same weight for higher frequencies opposite to lower frequencies.\n\nWhen disabled, the frequency band is processed 'as is', but the usual audio signal has 3 dB/octave falloff in the frequency domain and could be compared with the pink noise. So lower frequencies take more effect on the Gate rather than higher frequencies.\n\nOn the other hand the Sidechain Boost mode allows to compensate the -3 dB/octave falloff of the signal spectrum and, even more, makes the signal spectrum growing +3 dB/octave in the almost fully audible frequency range.\n\n**Pink** applies +3 dB/octave while **Brown** applies +6 dB/octave Sidechain Boost. Each of them can use bilinear-transformed (BT) or matched-transformed (MT) shelving filter.\n\n**Sidechain Source** \nIf the External Sidechain is enabled inside at least one band, this combobox allows to select the input device as source.\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal mixed in the output.\n\n## Band Options\n\n**Band Start** \nAllows to change the lower end split frequency of the selected band. This value is assigned to the Band End of the previous enabled band. It can be modified for all bands except the first one, which is always enabled and its value is 0 Hz.\n\n**Band End** \nSpecify the upper end split frequency of the selected band. It cannot be directly modified and assumes the same value of the Band Start of the next enabled band. For the last enabled band it is always 24.000 Hz.\n\n**External Sidechain** \nThe Sidechain is an external source took by a specific input device (typically a microphone).\n\n**Band Bypass** \nIf enabled, the selected band is not affected by the gating stage.\n\n**Solo** \nTurns on the Solo mode to the selected band by applying -36 dB gain to the other non-soloing bands.\n\n**Mute** \nTurns on the Mute mode applying -36 dB gain to to the selected band.\n\n**Attack Time** \nThe length of time it takes to apply roughly two-thirds of the gain reduction.\n\n**Release Time** \nThe length of time it takes to restore roughly two-thirds of the gain reduction.\n\n**Curve Threshold** \nThe Gate fully opens upon the Sidechain level becoming above Curve Threshold.\n\nIf Hysteresis is not enabled, the Gate begins to close upon the Sidechain level becoming below Curve Threshold.\n\n**Curve Zone Size** \nThe Gate begins to open upon the Sidechain level becoming above the Curve Threshold + Curve Zone.\n\nIf Hysteresis is not enabled, the Gate fully closes upon the Sidechain level becoming below the Curve Threshold + Curve Zone.\n\n**Hysteresis** \nWhen enabled, Curve Threshold and Curve Zone apply only to the opening Gate, and separate parameters can be configured for closing Gate.\n\n**Hysteresis Threshold** \nIf Hysteresis is enabled, the Gate begins to close upon the Sidechain level becoming below Curve Threshold + Hysteresis Threshold.\n\n**Hysteresis Zone Size** \nIf Hysteresis is enabled, the Gate fully closes upon the Sidechain level becoming below the Curve Threshold + Hysteresis Threshold + Hysteresis Zone.\n\n**Reduction** \nIf the value is negative, it acts as the amount of gain reduction to apply to the input signal when the Gate is fully closed. If the value is positive, the Gate operates in \"Reverse Mode\": It reduces the amplitude when the signal is above the threshold.\n\n**Makeup** \nThe gain to apply after the gating stage.\n\n## Band Sidechain Options\n\n**Mode** \nDetermines how the Sidechain is evaluated for the gating stage.\n\n- **Peak** - The Gate reacts according to the peaks.\n- **RMS** - The Gate reacts according to the average loudness measured by the root mean square.\n- **Low-Pass Filter** - The Gate reacts according to the signal processed by a recursive 1-pole Low-Pass filter.\n- **Simple Moving Average** - The Gate reacts according to the signal processed by the Simple Moving Average filter.\n\n**Source** \nDetermines which part of the Sidechain is taken into account for the gating stage.\n\n- **Middle** - The sum of left and right channels.\n- **Side** - The difference between left and right channels.\n- **Left** - Only left channel is used.\n- **Right** - Only right channel is used.\n- **Min** - The absolute minimum value is taken from stereo input.\n- **Max** - The absolute maximum value is taken from stereo input.\n\n**Low-Cut Filter** \nEnables a custom Low-Cut Filter for the selected band.\n\n**Low-Cut Frequency** \nSets the cut-off frequency of the custom Low-Cut filter. If it is disabled, the default Low-Cut filter assumes internally the Band Start frequency as the cut-off frequency.\n\n**Hight-Cut Filter** \nEnables a custom High-Cut Filter for the selected band.\n\n**Hight-Cut Frequency** \nSets the cut-off frequency of the custom High-Cut filter. If it is disabled, the default High-Cut filter assumes internally the Band End frequency as the cut-off frequency.\n\n**PreAmp** \nGain applied to the Sidechain signal of the selected band.\n\n**Reactivity** \nThe time that defines the number of samples used to process the Sidechain in RMS, Uniform and Low-Pass modes. Higher the value, more smooth the gating.\n\n**Lookahead** \nThe band signal to gate is delayed by this amount of time, so that the gating will be applied earlier than it would be otherwise.\n\nEach band can have different Lookahead values. To avoid phase distortions in the mixing stage, all the bands are automatically delayed for an individually calculated period of time.\n\n## References\n\n- [LSP Sidechain Multiband Gate Stereo](https://lsp-plug.in/?page=manuals§ion=sc_mb_gate_stereo)\n- [Wikipedia Noise Gate](https://en.wikipedia.org/wiki/Noise_gate)\n","isInternal":false,"tokens":1518,"sizeBytes":6916},{"name":"pitch.md","path":"src/contents/docs/plugins/pitch.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/pitch.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Pitch\n\nPitch shifting is a sound recording technique in which the original Pitch of a sound is raised or lowered. Easy Effects uses the pitch shifter from SoundTouch.\n\n**Mode** \nControls the method used for pitch shifting.\n\n- **High Speed** - Uses a method with a CPU cost that is relatively moderate and predictable.\n- **High Quality** - Uses the highest quality method for pitch shifting. This CPU cost is approximately proportional to the required frequency shift.\n- **High Consistency** - Uses the method that gives greatest consistency when used to create small variations in pitch around the 1.0-ratio level. Unlike the previous two options, this avoids discontinuities when moving across the 1.0 pitch scale. It also consumes more CPU than the others in the case where the pitch scale is exactly 1.0.\n\n**Formant** \nControls the handling of formant shape (spectral envelope) when pitch-shifting.\n\n- **Shifted** - Applies no special formant processing. The spectral envelope will be pitch shifted as normal.\n- **Preserved** - Preserves the spectral envelope of the unshifted signal. This permits shifting the note frequency without so substantially affecting the perceived pitch profile of the voice or instrument.\n\n**Transients** \nControls the component frequency phase-reset mechanism that may be used at transient points to provide clarity and realism to percussion and other significant transient sounds.\n\n- **Crisp** - Resets component phases at the peak of each transient (the start of a significant note or percussive event). This usually results in a clear-sounding output, but it is not always consistent, and may cause interruptions in stable sounds present at the same time as transient events.\n- **Mixed** - Resets component phases at the peak of each transient, outside a frequency range typical of musical fundamental frequencies. The results may be more regular for mixed stable and percussive notes than Crisp option, but with a \"phasier\" sound. The balance may sound very good for certain types of music and fairly bad for others.\n- **Smooth** - Does not reset component phases at any point. The results will be smoother and more regular but may be less clear than the other transient options.\n\n**Detector** \nControls the type of transient detector used.\n\n- **Compound** - Uses a general purpose transient detector which is likely to be good for most situations.\n- **Percussive** - Detects percussive transients.\n- **Soft** - Uses an onset detector with less of a bias toward percussive transients. This may give better results with certain material (e.g. relatively monophonic piano music).\n\n**Phase** \nControls the adjustment of component frequency phases from one analysis window to the next during non-transient segments.\n\n- **Laminar** - Adjusts phases when stretching in such a way as to try to retain the continuity of phase relationships between adjacent frequency bins whose phases are behaving in similar ways. This should give good results in most situations.\n- **Independent** - Adjusts the phase in each frequency bin independently from its neighbours. This usually results in a slightly softer, phasier sound.\n\n**Cents** \nNumber of cents the Pitch will be increased or decreased.\n\n**Semitones** \nNumber of semitones the Pitch will be increased or decreased.\n\n**Octaves** \nNumber of octaves the Pitch will be increased or decreased.\n\n## References\n\n- [Wikipedia Pitch Shift](https://en.wikipedia.org/wiki/Pitch_shift)\n- [SoundTouch Audio Time Stretcher Library - Attractive Features](https://www.surina.net/soundtouch/)\n","isInternal":false,"tokens":737,"sizeBytes":3565},{"name":"reverb.md","path":"src/contents/docs/plugins/reverb.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/reverb.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Reverberation\n\nReverberation is the phenomenon of persistence of sound after the source has been stopped as a result of multiple reflections of the waves over objects within a closed surface. These reflections build up with each other and decay gradually as they are absorbed by the surfaces of objects in the space enclosed.\n\nThe Reverberation is different than Echo because the Echo is a reflected sound wave with sufficient magnitude and delay to be detectable as a signal distinct from the source one. To simulate the Reverberation effect, Easy Effects uses the Reverb plugin developed by Calf Studio Gear.\n\n**High Frequency Damping** \nCutoff frequency of the reflections. It causes higher frequencies to decay faster.\n\n**Room Size** \nSize of the space where simulated Reverberation occurs inside. It determines the time between reflections.\n\n**Diffusion** \nDegree of uniformity. Higher values lead to less uniform Reverberation.\n\n**Pre Delay** \nAdditional delay. It corresponds to a distance between sound source and the nearest wall.\n\n**Decay Time** \nThe time it takes for Reverberation to fade out.\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal (Reverberation) mixed in the output.\n\n**Bass Cut** \nRemoves low frequencies from the Reverberation.\n\n**Treble Cut** \nRemoves high frequencies from the Reverberation.\n\n## References\n\n- [Wikipedia Reverberation](https://en.wikipedia.org/wiki/Reverberation)\n- [Calf Reverb](https://calf-studio-gear.org/doc/Reverb.html)\n","isInternal":false,"tokens":336,"sizeBytes":1543},{"name":"rnnoise.md","path":"src/contents/docs/plugins/rnnoise.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/rnnoise.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Noise Reduction\n\nThe Noise Reduction is a process aimed to attenuate the disturbing noise from a signal.\n\nEasy Effects Noise Reduction is made on the RNNoise library which is based based on recurrent neural network, a class of artificial neural networks where connections between nodes form a directed graph along a temporal sequence. This allows it to exhibit temporal dynamic behavior.\n\nStandard RNNoise Model is used and custom models can be imported to perform different types of noise reduction.\n\n## References\n\n- [Wikipedia Noise Reduction](https://en.wikipedia.org/wiki/Noise_reduction)\n- [Wikipedia Recurrent Neural Network](https://en.wikipedia.org/wiki/Recurrent_neural_network)\n- [Jean-Marc Valin - RNNoise: Learning Noise Suppression](https://jmvalin.ca/demo/rnnoise/)\n","isInternal":false,"tokens":158,"sizeBytes":783},{"name":"spectrum.md","path":"src/contents/docs/plugins/spectrum.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/spectrum.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Spectrum\n\n**Enabled** \nShow or hide the Spectrum.\n\n**Shape** \nChoose the style of the animation.\n\n**Points** \nNumber of points/bars on the x axis.\n\n**Height** \nChanges the height of the Spectrum.\n\n**Line Width** \nThe width of a single bar.\n\n**Fill** \nDraw filled bars.\n\n**Show Bars Border** \nDraw bars with borders.\n\n**Rounded Corners** \nDraw bars with rounded corners.\n\n**Color** \nChoose the color of the Spectrum components.\n\n**Minimum Frequency Range** \nLower end frequency of the Spectrum.\n\n**Maximum Frequency Range** \nUpper end frequency of the Spectrum.\n\n[Settings Menu](../user_interface/settingsmenu.md)","isInternal":false,"tokens":142,"sizeBytes":625},{"name":"speex.md","path":"src/contents/docs/plugins/speex.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/speex.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Speech Processor\n\nThis plugin allows EasyEffects to use the Speex preprocessor to attenuate disturbing background noises from a signal.\n\nCompared to Noise Reduction which uses RNNoise to suppress noises, Speech Processor has the benefit of using less computational resources, at the cost of sacrificing noise suppression quality.\n\nFor more information on noise suppression in general, refer to the manual page on Noise Reduction.\n\n## References\n\n- [The Speex Project](https://www.speex.org/)\n","isInternal":false,"tokens":95,"sizeBytes":494},{"name":"stereotools.md","path":"src/contents/docs/plugins/stereotools.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/stereotools.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Stereo Tools\n\nEasy Effects uses the Stereo Tools developed by Calf Studio Gear which provides some handy utilities to manage Stereo streams handling Left and Right channels in conjunction with Mid and Side signals.\n\nMid/Side is a recording technique which registers Mid and Side signals rather then plain Left and Right channels. Its basic setup makes use of one cardioid microphone (Mid) and one bidirectional (figure-eight) microphone (Sides). In mastering stage Mid and Side can be used to shape the balance of the Stereo image having more control over the mix to get a wider, deeper, and more focused track.\n\nHaving a Stereo stream, Mid is obtained summing both channels `(L+R)` while Side is the result of the difference between Left and Right `(L-R)`. Mid and Side signals can be decoded back to Left `(M+S)/2` and Right `(M-S)/2`.\n\n## Input\n\n**Balance** \nSets the balance between both channels.\n\n**Softclip (button)** \nMakes a kind of analog distortion instead of harsh digital 0 dB clipping.\n\n**Softclip (value)** \nLevel of Softclip.\n\n## Stereo Matrix\n\n**Mode** \nAllows to choose between different channel configurations such as normal Stereo Mode, Mid-Side or Mono simulation.\n\n**Mute L/R** \nMutes the Left/Right channel.\n\n**Invert Phase L/R** \nChanges the Phase of the Left/Right channel.\n\n**Side Level** \nThe level of the Side signal.\n\n**Side Balance** \nThe balance of the Side signal.\n\n**Middle Level** \nThe level of the Middle signal.\n\n**Middle Panorama** \nThe position in the Panorama of the Middle signal.\n\n## Output\n\n**Balance** \nSets the balance between both channels.\n\n**Delay L/R** \nDelays the Left or the Right channel. Negative values delay the Left channel and positive values the Right channel.\n\n**Stereo Base** \nSets the Stereo Base of the content seamless between Mono and inverted channels.\n\n**Stereo Phase** \nSet the Stereo Phase of the content.\n\n## References\n\n- [Wikipedia Stereophonic Sound](https://en.wikipedia.org/wiki/Stereophonic_sound)\n- [Calf Stereo Tools](https://calf-studio-gear.org/doc/Stereo%20Tools.html)\n- [Unlock Your Sound - Stereo, Mono, Mid, Side, Panning, and Imaging Explained](https://unlockyoursound.com/stereo/)\n","isInternal":false,"tokens":498,"sizeBytes":2181},{"name":"voicesuppressor.md","path":"src/contents/docs/plugins/voicesuppressor.md","rawUrl":"https://raw.githubusercontent.com/wwmm/easyeffects/HEAD/src/contents/docs/plugins/voicesuppressor.md","title":"Plugins Skill","category":"anthropic-skill","format":"markdown","content":"# Voice Suppressor\n\nVery simple plugin that removes voice from audio. It calculates the correlation of the lefft and right channels in\nthe frequency domain. Frequencies whose correlation and phase difference are beyond the configured thresholds are attenuated. \n\n**Start** \n\nFrequencies below this value are not attenuated. \n\n**End** \n\nFrequencies above this value value are not attenuated. \n\n**Correlation** \n\nFrequencies are attenuated only if the correlation between left and right channel is at least equal to this value. \n\n**Phase Difference** \n\nFrequencies are not attenuated when the magnitude of phase difference between left and right channel is above this value. \n\n**Minimum Kurtosis** \n\nFrequencies with local kurtosis below this value are attenuated. \n\n**Maximum Instantaneous Frequency** \n\nThis is related to the frequency of rotation of the complex vector that represents the correlation between the left and right channel at a given frequency.\nIn other words this measures how fast the phase difference between the left and right channels changes at a given frequency bin in the Fourier transform.\nMore attenuation is applied to the channels samples that have rotation frequency below this parameter value. \n\n**Inverted Mode** \nInstead of suppressing voice the plugin will try to suppress the background and keep the voice. \n","isInternal":false,"tokens":259,"sizeBytes":1355}],"systemPromptSnippet":"\n\n# Auto Gain\n\nEasy Effects Autogain is based on the library libebur128 which implements the EBU R 128 standard for loudness normalization. It changes the audio volume to a perceived loudness target that can be customized by the user.\n\n**Target** \nLoudness level.\n\n**Silence** \nSilence threshold. While the momentary loudness is below this value the autogain won't make any changes to the current gain being applied to the signal level.\n\n**Maximum History** \nRange of time taken into account for the calculation of loudness level and output gain.\n\n**Reference** \nThe parameter used as reference to evaluate the output gain.\n\n- **Momentary** - Measures the loudness of the past 400 milliseconds.\n- **Short-Term** - Measures the loudness of the past 3 seconds.\n- **Integrated** - Ideally it indicates how loud the content is on average. It measures the loudness on a long range of time, depending on the value set as Maximum History.\n- **Geometric Mean** - Uses the geometric mean of all the above mentioned parameters, or two of them at user choosing.\n\n**Reset History** \nResets the Autogain history related to chosen Reference.\n\n**Monitor Parameters** \nAutogain values shown as stats.\n\n- **Relative** - Used to detect silence. Whenever the Momentary term is below a predetermined threshold, modifications to the output gain will be disabled.\n- **Loudness** - The difference between its value and the target loudness determines the output gain.\n- **Range** - Indicates how large is the dynamic range of the content played.\n- **Output Gain** - The input signal is adjusted by this correction gain to bring its loudness to the target value.\n\n## References\n\n- [Wikipedia EBU R 128](https://en.wikipedia.org/wiki/EBU_R_128)\n- [EBU - Loudness Normalisation and Permitted Maximum Level of Audio Signals](https://tech.ebu.ch/publications/r128/)\n\n\n# Autotune\n\nAutotune is a pitch correction effect that automatically adjusts the pitch of an audio signal to the nearest target note. It can be used to subtly correct vocal intonation or to create the characteristic \"auto-tune effect\" heard in modern music. Easy Effects uses the fat1 plugin from x42 LV2 Plugins.\n\n## Tuning Options\n\n**Mode** \nControls how pitch correction is applied.\n\n- **Auto** - Automatically detects the pitch and corrects it to the nearest enabled note in the scale.\n- **Manual** - Targets a fixed pitch set by the Offset parameter.\n\n**Tuning** \nThe reference frequency for the A4 note, in Hz. The standard tuning is 440 Hz. Adjust this if you are working with non-standard tuning (range: 400–480 Hz).\n\n**Correction** \nControls the strength of pitch correction. At 1.0, the pitch is fully corrected to the target note. Lower values allow more of the natural pitch variation through, producing a subtler effect.\n\n**Bias** \nControls the preference toward the current note versus the nearest target note. At 0.5 (center), no bias is applied. Higher values make the correction stick to the current note longer before switching. Lower values make it switch to the nearest target more readily.\n\n**Filter** \nControls the smoothness of the pitch detection. Lower values result in faster, more responsive detection but may introduce artifacts. Higher values produce smoother, more stable detection but with slower response.\n\n**Offset** \nShifts the target pitch by the specified number of semitones (range: -2.0 to 2.0). In Manual mode, this sets the fixed target pitch offset.\n\n**Fast Correction** \nWhen enabled, applies pitch correction with minimal latency. This produces a more immediate, aggressive correction effect but may introduce more artifacts.\n\n## Scale Selection\n\nThe Scale section allows you to select which notes the pitch correction will target.\n\n**Root** \nThe root note of the scale (C, D, E, F, G, A, or B).\n\n**Accidental** \nApplies a sharp or flat modifier to the root note.\n\n**Scale** \nSelects either Major or Minor scale pattern. Choosing a scale automatically enables the correct notes for that key.\n\n**Note Toggles** \nIndividual toggles for each of the 12 chromatic notes (C through B). Selecting a scale sets these automatically, but you can also configure them manually for custom scales or chromatic correction (all notes enabled).\n\n## Pitch Error Meter\n\nDisplays the current pitch deviation from the target note. A value of 0 means the pitch is exactly on target. Negative values indicate the pitch is flat, positive values indicate it is sharp.\n\n## References\n\n- [x42 fat1 Auto Tune](https://x42-plugins.com/x42/x42-autotune)\n- [Wikipedia Auto-Tune](https://en.wikipedia.org/wiki/Auto-Tune)\n\n\n# Bass Enhancer\n\nA Bass Enhancer is used to produce very low sound that is not present in the original signal. This is done by creating harmonic distortions of the signal which are restricted in range and added to the original signal. It raises the lower end of an audio signal without simply raising the lower frequencies like an [Equalizer](equalizer.md) would do to create a more \"fat\" or \"boomy\" sound.\n\n**Amount** \nAmount of harmonics added to the original signal.\n\n**Harmonics** \nAmount of newly created harmonics.\n\n**Scope** \nThe frequency above which harmonics are produced.\n\n**Floor (button)** \nConstrain the enhancement on the lower end.\n\n**Floor (value)** \nThe frequency below which no harmonics are produced.\n\n**Blend Harmonics** \nThe \"colour\" (or octave) of the harmonics.\n\n**Listen** \nMute the original signal and listen to the harmonics exclusively.\n\n## References\n\n- [Calf Bass Enhancer](https://calf-studio-gear.org/doc/Bass%20Enhancer.html)\n- [Wikipedia Missing Fundamental](https://en.wikipedia.org/wiki/Missing_fundamental)\n\n\n# Bass Loudness\n\nThe ear is less sensitive to low frequencies when listening at low volume. This plugin developed by MDA allows the bass level to be adjusted measuring the \"equal-loudness contour\".\n\n**Loudness** \nSource level relative to listening level (based on a 100 dB SPL maximum level).\n\n**Output** \nChange output level.\n\n**Link** \nAutomatically adjusts Output to maintain a consistent tonal balance at all levels.\n\n## References\n\n- [Wikipedia Equal-Loudness Contour](https://en.wikipedia.org/wiki/Equal-loudness_contour)\n- [MDA Loudness](http://mda.smartelectronix.com/vst/help/loudness.htm)\n\n\n# Compressor\n\nA Compressor is used to reduce the dynamic range or, in other words, the difference in level between the quietest and the loudest parts of an audio signal. It achieves this purpose altering the gain when the signal overtakes a predetermined Threshold. Easy Effects uses the Stereo Sidechain Compressor from Linux Studio Plugins.\n\n## Compressor Options\n\n**Attack Time** \nThe length of time it takes to apply roughly two-thirds of the targeted amount of compression ratio to the uncompressed signal.\n\n**Release Time** \nThe length of time it takes to restore roughly two-thirds of the reduced gain (in Downward mode) or increased gain (in Upward/Boosting mode) to the compressed signal.\n\n**Attack Threshold** \nThe target level around which the compression is applied (the range depends by the Knee).\n\n**Release Threshold** \nSets up the Threshold of the Release Time, calculated by summing the Release Threshold to the Attack Threshold.\n\nIf the Sidechain level is above that Threshold, the compressor uses the Release Time for the releasing stage. Otherwise the Attack Time is used in place of Release Time.\n\nFor example, with -10 dB Attack Threshold and -60 dB Release Threshold, if the Sidechain is above `-10 + (-60) = -70 dB`, the Release Time is used for the gain restoration. If the Sidechain is below -70 dB, the Attack Time is used for the gain restoration.\n\n**Ratio** \nThe amount of attenuation (in Downward mode) or amplification (in Upward/Boosting mode) that will be applied to the signal.\n\nFor example, when the Ratio is 2 in Downward mode and the Sidechain rises above the Threshold by 10 dB, the signal would be ideally reduced by 5 dB (10/2 dB). In practice this behavior mostly depends on how the Compressor is designed and configured.\n\n**Knee** \nThe range over which the Compressor switches from no compression to almost the full ratio compression (the Threshold usually sits at the center of this transition zone).\n\n**Makeup** \nThe gain to apply after the compression stage.\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal mixed in the output.\n\n**Mode** \n- **Downward** - It's aimed to decrease the gain of the signal above the Threshold.\n- **Upward** - It's aimed to increase the gain of the signal below the Threshold.\n- **Boosting** - It's aimed to increase the gain of the signal below the Threshold by a specific amount.\n\n**Boost Threshold** \nThe Threshold below which a constant amplification will be applied to the input signal in Upward Mode (it prevents from applying infinite amplification to very quiet signals).\n\n**Boost Amount** \nMaximum gain amplification to apply in Boosting Mode.\n\n## Sidechain\n\n**Listen** \nAllows to listen the processed Sidechain signal.\n\n**Input Type** \nDetermines which signal is the Sidechain or, in other words, the signal that controls the compression stage.\n\n- **Feed-forward** - The Sidechain is the Compressor input signal (taken after applying the plugin input gain). More aggressive compression.\n- **Feed-back** - The Sidechain is the Compressor output signal (taken before applying the Makeup and the plugin output gain). Vintage-style compression.\n- **External** - The Sidechain is an external source took by a specific input device (typically a microphone).\n\n**Input Device** \nSelect the device for the External Sidechain.\n\n**Mode** \nDetermines how the Sidechain is evaluated for the compression stage.\n\n- **Peak** - The Compressor reacts according to the peaks.\n- **RMS** - The Compressor reacts according to the average loudness measured by the root mean square.\n- **Low-Pass** - The Compressor reacts according to the signal processed by a Low-Pass filter.\n- **Uniform** - The Compressor reacts according to the loudness measured by the average of the absolute amplitude.\n\n**Source** \nDetermines which part of the Sidechain is taken into account for the compression stage.\n\n- **Middle** - The sum of left and right channels.\n- **Side** - The difference between left and right channels.\n- **Left** - Only left channel is used.\n- **Right** - Only right channel is used.\n- **Min** - The absolute minimum value is taken from stereo input.\n- **Max** - The absolute maximum value is taken from stereo input.\n\n**PreAmplification** \nGain applied to the Sidechain signal.\n\n**Reactivity** \nThe time that defines the number of samples used to process the Sidechain in RMS, Uniform and Low-Pass modes. Higher the value, more smooth the compression.\n\n**Lookahead** \nThe signal to compress is delayed by this amount of time, so that the compression will be applied earlier than it would be otherwise. The corresponding delay is reproduced on the output signal.\n\n## Sidechain Filters\n\n**High-Pass Filter Mode** \nSets the type of the High-Pass filter applied to Sidechain signal.\n\n**High-Pass Frequency** \nSets the cut-off frequency of the High-Pass filter.\n\n**Low-Pass Filter Mode** \nSets the type of the Low-Pass filter applied to Sidechain signal.\n\n**Low-Pass Frequency** \nSets the cut-off frequency of the Low-Pass filter.\n\n## References\n\n- [Wikipedia Dynamic Range Compression](https://en.wikipedia.org/wiki/Dynamic_range_compression)\n- [LSP Sidechain Compressor Stereo](https://lsp-plug.in/?page=manuals§ion=sc_compressor_stereo)\n- [Black Ghost Audio - The Ultimate Guide to Compression](https://www.blackghostaudio.com/blog/the-ultimate-guide-to-compression)\n- [Attack Magazine - Demolishing The Myths of Compression](https://www.attackmagazine.com/features/columns/gregory-scott-demolishing-the-myths-of-compression/)\n\n\n# Convolver\n\nThe Convolver creates a simulation of an audio environment using a pre-recorded audio sample of the impulse response of the space being modeled. This feature is based on the \"convolution\": a process through which the sonic characteristics of one signal are used to alter the character of another.\n\nEasy Effects Convolver offers the opportunity to apply multiple impulse responses by combining them in one file.\n\n**Impulses** \nManage impulse response files. This menu allows to load a specific impulse or add/remove files from the Easy Effects configuration directory.\n\n**Combine** \nSelect two impulse responses to combine into a single file to be saved under the Easy Effects configuration directory.\n\n**Stereo Width** \nModify the impulse response stereo image width.\n\n**Spectrum** \nVisualize the frequency spectrum of the selected channel.\n\n## References\n\n- [Wikipedia Reverb Effect](https://en.wikipedia.org/wiki/Reverb_effect)\n- [Indiana University - Convolution: A Form of Cross-Synthesis](https://cmtext.indiana.edu/synthesis/chapter4_convolution.php)\n- [Designing Sound - Recording Impulse Responses](https://designingsound.org/2012/12/29/recording-impulse-responses/)\n\n\n# Crossfeed\n\nEasy Effects uses the Crossfeed by bs2b. This plugin is used to improve headphone listening of stereo audio records. It does so by mixing the left and right channel in a way that simulates a stereo speaker setup while using headphones.\n\n**Cutoff** \nLow-Pass filter cutoff frequency.\n\n**Feed** \nAmount of signal from a channel that is sent to the other.\n\n**Presets** \n- **Default** - Closest to virtual speaker placement (30°, 3 meters).\n- **Cmoy** - Close to Chu Moy's Crossfeed (popular).\n- **Jmeier** - Close to Jan Meier's CORDA amplifiers (little change).\n\n## References\n\n- [Wikipedia Crossfeed](https://en.wikipedia.org/wiki/Crossfeed)\n- [Bauer Stereophonic-to-Binaural DSP.](http://bs2b.sourceforge.net/)\n\n\n# Crosstalk Canceller\n\nWhen audio is played by a stereo speaker audio from both channels arrive to the listener ears. As the superposition\nof each channel audio will happen with a delay that depends on the distance between the listener ear and the speakers\nundesirable effects can happen. That superposition is called crosstalk in this context and can be attenuated by this\nplugin.\n\n\n# Crusher\n\nA bitcrusher reduces the resolution or bandwidth of digital audio data. Audio reduced in bit depth sounds more harsh and \"digital\". \nIn the bitcrusher from Calf Studio Gear used in EasyEffects reduction can be done in a linear or logarithmic way.\nAccording to the plugin authors the logarithmic way results in a much smoother sound in low volume signals. \n\n**Mode** \n- **Linear** - Linear distance between bits.\n- **Logarithmic** - Logarithmic distances between bits. The result is a much more \"natural\" sounding crusher which doesn't gate low signals. \n\n**Bit reduction** \nControls the processed audio bit depth. \n\n**DC offset** \nThis offset causes different crushing of the lower and the upper half of the signal. \n\n**Anti-aliasing** \nControls the softness of the crushing sounds. \n\n**Mix** \nControls the mix of original and processed audio. When at 0% the output signal is equal to the original. When at 100% only the processed signal is in the output signal. \n\n**Reduction** \nControls the reduction of the sample rate (downsampling). \n\n**Low frequency oscillator active** \nEnable the low frequency oscillator. The oscillator will make the sample rate reduction change between a range determined by the range control.\n\n**Low frequency oscillator range** \nControls the amount of modulation applied to the sample rate reduction. \n\n**Low frequency oscillator rate** \nControls the frequency of the oscillator. \n\n## References\n\n- [Wikipedia Bit Crusher](https://en.wikipedia.org/wiki/Bitcrusher)\n- [Calf Crusher](https://calf-studio-gear.org)\n- [Calf Crusher Documentation](https://calf-studio-gear.org/doc/Crusher.html)\n\n\n# Crystalizer\n\nThe Crystalizer plugin can be used to add a little of dynamic range to songs that were overly compressed. The signal is split in multiple bands to which different intensities can be applied in order to alter the overall dynamic range.\n\n**Intensities**\nThe higher the value the higher is the difference in magnitude between the loudest and the quietest sounds of the selected band. Different intensities can be set for each frequency band.\n\n**Bypass**\nWhen active the audio signal passing through the selected band is not modified.\n\n**Mute**\nMutes the selected band.\n\n**Adaptive Intensity**\nWhen enabled the intensity value set by the band slider is continuously scaled as audio is processed. The scaling may\nincrease or decrease the intensity based on the geometric mean of the signal crest factor, kurtosis and spectral flux.\n\nWhen using static intensity values it may happen that the crystalizer enhances high frequency audio too much. What leads\nto audible artifacts. The adaptive scaling removes this is most of the cases.\n\n**Oversampling**\nResamples the signal to double its sampling rate value and back to the original value after the plugins effects are applied.\nIt improves the effects quality but it also increases CPU usage.\n\n**Oversampling Quality**\nControls the resampling quality. Higher values require more CPU power.\n\n**Fixed Quantum**\nForces the plugin to use a fixed audio buffer size to process audio. This helps to avoid noises when the sound server switches latency on the fly.\nBut it has the downside of introducing unneeded latency in some cases. And increased CPU usage in cases where higher latency could be used.\n\n**Transition Band**\nControls the width of the transition band of the bandpass filters used to split the signal in several bands. Higher values will use less CPU. But will result in more overlap between the bands.\n\n## References\n\n- [Wikipedia Dynamic Range](https://en.wikipedia.org/wiki/Dynamic_range)\n- [Crest factor](https://en.wikipedia.org/wiki/Crest_factor)\n- [Kurtosis](https://en.wikipedia.org/wiki/Kurtosis)\n- [Spectral flux](https://en.wikipedia.org/wiki/Spectral_flux)\n\n\n# Deep Noise Remover\n\nAdvanced noise reduction is achieved using \"DeepFilterNet\" technology. This technology uses AI technology called deep learning to efficiently remove noise from audio signals, enabling higher quality noise suppression than conventional methods. This technology has several control parameters, which are explained below and how to set them.\n\n**Model** \nSelects the noise reduction model. Each one must be installed separately.\n\n- DeepFilterNet: requires [DeepFilterNet](https://github.com/Rikorose/DeepFilterNet).\n- DPDFNet: requires [HushMic](https://github.com/Fovty/HushMic). The Attenuation Limit and Process as Mono controls apply to this model.\n\n**Process as Mono (DPDFNet only)**\n\nAverages the left and right input channels before resampling, processes the result with one model instance, and sends the same processed signal to both output channels. This can significantly reduce CPU usage for mono microphones that supply the same signal on both channels. It does not change the selected model or its noise reduction quality.\n\nThis setting is off by default and is saved in presets. Enable it for mono speech input; leave it off when stereo separation matters. If only one input channel contains audio, averaging reduces its level by 6 dB. Opposite-polarity signals can cancel, so check the input channels before enabling it.\n\n**Attenuation Limit** \nThis parameter determines how much noise is attenuated. Higher values result in stronger noise reduction, but too high a value may also remove parts of the audio.\n\n- Recommended setting: 70dB to 80dB is a typical balanced value. This minimizes background noise while preserving speech intelligibility.\n\n**Minimum Processing Threshold** \nSignals below the Minimum Processing Threshold are targeted for denoising. Signals louder than this threshold are not subject to noise reduction and are passed through unchanged. This setting is intended to effectively remove small background noise.\n\n- Recommended setting: -15dB to -30dB is good. This should be fine-tuned according to the ambient noise. Currently, Easy Effects does not allow a value smaller than -15dB.\n\n**Maximum ERB Processing Threshold** \nThe human auditory system has the ability to separate and recognize different frequency components. Equivalent Rectangular Bandwidth (ERB) is a numerical expression and model of this filtering capability. ERB-based processing emphasizes the perceptual importance of speech by focusing on specific frequency bands. This improves the effectiveness of noise reduction and speech enhancement. This threshold sets the maximum level at which perceptually equivalent bandwidth (ERB)-based processing is applied. Signals above this threshold are not subject to noise reduction or speech enhancement processing. This prevents important speech components or loud portions of speech from being unnecessarily processed.\n\n- Recommended setting: Start in the range of 20dB to 30dB and adjust as needed.\n\n**Maximum DF Processing Threshold** \nThe Maximum DF Processing Threshold sets the maximum level of signal that DeepFilterNet will process. Signals exceeding this threshold are excluded from noise reduction processing. This prevents important or loud portions of speech from being unnecessarily processed.\n\n- Recommended setting: 20 dB as the default value. This effectively suppresses noise while maintaining voice quality.\n\n**Minimum Processing Buffer** \nDefines the size of the buffer used to process frames of audio data. A larger buffer can improve noise suppression but may increase latency.\n\n- Recommended setting: Set to a small value (0) if real-time processing is important, or to a number greater than 1 if noise suppression accuracy is a priority.\n\n**Post Filter Beta** \nControls the intensity of the post-processing filter applied after the initial noise suppression. This allows for more subtle refinement of the audio signal.\n\n- Recommended setting: 0.5dB to 2dB is generally recommended, but currently Easy Effects has a maximum value of 0.05dB, which is almost ineffective.\n\n\n# Deesser\n\nA Deesser is used to dynamically reduce high frequencies. The standard field of use of this plugin is the reduction of \"sssss\" and \"shhhh\" in vocal tracks. Easy Effects uses the Deesser developed by Calf Studio Gear.\n\n**Detection** \nSelect the detection of the Sidechain signal between Peak (stronger) and RMS (smoother).\n\n**Mode** \nSelect the operation mode between Wideband and Split. In Split mode not the full range signal will be affected by the gain reduction, but only frequencies above the split frequency will be manipulated in gain.\n\n**F1 Split** \nThe split frequency. All signals above this frequency will affect the gain reduction (and are affected in Split mode too).\n\n**F1 Gain** \nIt shifts the volume of the higher band. In Wideband mode it affects the Sidechain. In Split mode it also affects the processed high frequencies.\n\n**F2 Peak** \nCenter frequency of the bell filter. It allows a more precise selection of the Sidechain signal.\n\n**F2 Level** \nIncreases or decreases the level of the chosen F2 frequency.\n\n**F2 Peak Q** \nSet the quality of the bell filter. Higher values will affect a narrower frequency range. Lower values will affect a wider band.\n\n**Laxity** \nThe reaction of the Deesser. Higher values won't affect really short peaks.\n\n**Threshold** \nThe level above which the gain reduction is applied.\n\n**Ratio** \nThe amount of attenuation applied to the signal.\n\nFor example, a Ratio of 2 means that if the level rises 4 dB above the Threshold, it will be only 2 dB above after the reduction.\n\n**Makeup** \nThe gain to apply after the processing stage. In Split mode only the high band will be made up.\n\n## References\n\n- [Wikipedia De-essing](https://en.wikipedia.org/wiki/De-essing)\n- [Calf Deesser](https://calf-studio-gear.org/doc/Deesser.html)\n- [LedgerNote - De-esser: The Guide for Sibilant-Free Vocal Recordings](https://ledgernote.com/columns/mixing-mastering/de-esser/)\n\n\n# Delay\n\nThis plugin allows the user to add a short Delay of time to each individual channel of the stereo stream. Easy Effects Delay uses the Delay Compensator Stereo by Linux Studio Plugins set in time mode.\n\n**Left Delay** \nLeft channel delay in milliseconds.\n\n**Right Delay** \nRight channel delay in milliseconds.\n\n**Left Dry Level** \nAmount of right unprocessed signal mixed in the output.\n\n**Right Dry Level** \nAmount of right unprocessed signal mixed in the output.\n\n**Left Wet Level** \nAmount of left processed signal mixed in the output.\n\n**Right Wet Level** \nAmount of right processed signal mixed in the output.\n\n## References\n\n- [Wikipedia Delay (audio effect)](https://en.wikipedia.org/wiki/Delay_(audio_effect))\n- [LSP Delay Compensator Stereo](http://lsp-plug.in/?page=manuals§ion=comp_delay_x2_stereo)\n\n\n# Echo Canceller\n\nThe Echo is a reflected sound wave with sufficient magnitude and delay to be detectable as a signal distinct from the source one. An Echo Canceller is used to improve voice quality by preventing Echo from being created or removing it after it has been added to the source signal. Easy Effects uses the Echo Canceller from SpeexDSP library.\n\n**Frame Size** \nThe amount of time in milliseconds to process at once. It is recommended to use a frame size in the order of 20 ms.\n\n**Filter Length** \nThe amount of time of the Echo cancelling filter to use (also known as tail length). The recommended tail length is approximately the third of the room reverberation time. For example, in a small room, reverberation time is in the order of 300 ms, so a tail length of 100 ms is a good choice.\n\n## References\n\n- [Wikipedia Echo Suppression and Cancellation](https://en.wikipedia.org/wiki/Echo_suppression_and_cancellation)\n- [Speex Acoustic Echo Canceller](https://www.speex.org/docs/manual/speex-manual/node4.html#SECTION00450000000000000000)\n\n\n# Equalizer\n\nThe Equalization in sound recording and reproduction is the process of adjusting the volume of different frequency bands within an audio signal. Easy Effects uses the Parametric Equalizer from Linux Studio Plugins. The user can choose from 1 to 32 bands. Width and center frequency of each band can be customized as needed.\n\n## Global Options\n\n**Bands** \nThe number of bands.\n\n**Mode** \n- **IIR** - Infinite Impulse Response filters, nonlinear minimal phase. In most cases does not add noticeable latency to output signal.\n- **FIR** - Finite Impulse Response filters with linear phase, finite approximation of Equalizer's impulse response. Adds noticeable latency to output signal.\n- **FFT** - Fast Fourier Transform approximation of the frequency chart, linear phase. Adds noticeable latency to output signal.\n\n**Balance** \nBalance between left and right output channels.\n\n**Pitch Left** \nThe frequency shift for all filters of the left channel, in semitones.\n\n**Pitch Right** \nThe frequency shift for all filters of the right channel, in semitones.\n\n**Split Channels** \nWhen enabled it is possible to apply different configurations to left and right channels.\n\n**Flat Response** \nThis function sets each band gain to 0.\n\n**Calculate Frequencies** \nThis function calculates the center frequency and the width of each band using the current number of bands. Useful when the user wants fewer than 32 bands but has no idea about which frequencies should be chosen.\n\n## Band Options\n\n**Type** \n- **Off** - The filter is not working (turned off).\n- **Bell** - Bell filter with smooth peak/recess.\n- **High Pass** - High Pass filter with rejection of low frequencies.\n- **High Shelf** - Shelving filter with adjustment of high frequency range.\n- **Low Pass** - Low Pass filter with rejection of high frequencies.\n- **Low Shelf** - Shelving filter with adjustment of low frequency range.\n- **Notch** - Notch filter with full rejection of selected frequency.\n- **Resonance** - Resonance filter with sharp peak/recess.\n- **All Pass** - All Pass filter.\n\n**Mode** \n- **RLC** - Very smooth filters based on similar cascades of RLC contours. Bilinear Z-transform (BT) or Matched Z-transform (MT) is used for pole/zero mapping.\n- **BWC** - Butterworth-Chebyshev-type-1 based filters. Does not affect Resonance and Notch filters. Bilinear Z-transform (BT) or Matched Z-transform (MT) is used for pole/zero mapping.\n- **LRX** - Linkwitz-Riley based filters. Does not affect Resonance and Notch filters. Bilinear Z-transform (BT) or Matched Z-transform (MT) is used for pole/zero mapping.\n- **APO** - Digital biquad filters derived from canonic analog biquad prototypes digitalized through Bilinear transform. These are [textbook filters](https://shepazu.github.io/Audio-EQ-Cookbook/audio-eq-cookbook.html) which are implemented as in the [Equalizer APO](https://equalizerapo.com/) software. Direct design (DR) is used to serve the digital filter coefficients directly in the digital domain, without performing transforms.\n\n**Slope** \nThe slope of the filter characteristics.\n\n**Solo** \nMakes the selected band the only one active.\n\n**Mute** \nMutes the selected band.\n\n**Frequency** \nCenter frequency of the selected band.\n\n**Width** \nBandwidth calculated as `width = frequency / quality`.\n\n**Quality** \nThe quality factor of the filter used.\n\n## References\n\n- [Wikipedia Equalization (audio)](https://en.wikipedia.org/wiki/Equalization_(audio))\n- [LSP Parametric Equalizer x32 LeftRight](http://lsp-plug.in/?page=manuals§ion=para_equalizer_x32_lr)\n- [Wikipedia Q Factor](https://en.wikipedia.org/wiki/Q_factor)\n- [How to EQ - Q Factor and Bandwidth in EQ: What They Mean](https://howtoeq.wordpress.com/2010/10/07/q-factor-and-bandwidth-in-eq-what-it-all-means/)\n\n\n# Exciter\n\nAn Exciter is used to produce high sound that is not present in the original signal. This is done by creating harmonic distortions of the signal which are restricted in range and added to the original signal. It raises the upper end of an audio signal without simply raising the higher frequencies like an [Equalizer](equalizer.md) would do to create a more \"crisp\" or \"brilliant\" sound.\n\n**Amount** \nAmount of harmonics added to the original signal.\n\n**Harmonics** \nAmount of newly created harmonics.\n\n**Scope** \nThe frequency above which harmonics are produced.\n\n**Ceiling (button)** \nConstrain the excitement on the upper end.\n\n**Ceiling (value)** \nThe frequency above which no harmonics are produced.\n\n**Blend Harmonics** \nThe \"colour\" (or octave) of the harmonics.\n\n**Listen** \nMute the original signal and listen to the harmonics exclusively.\n\n## References\n\n- [Calf Exciter](https://calf-studio-gear.org/doc/Exciter.html)\n- [Wikipedia Exciter (effect)](https://en.wikipedia.org/wiki/Exciter_(effect))\n\n\n# Expander\n\nAn expander is an effect that works inversely to a compressor, further reducing the volume of quiet sounds below a certain level. This allows for natural suppression of background noise and unwanted whispers. For example, it is effective when air conditioning or other environmental sounds intrude into quiet scenes.\n\nThis effect is similar to the gate, which is explained in another chapter, in that it blocks quiet sounds. However, while a gate completely cuts sound, an expander gradually reduces sound, resulting in a more natural and smooth decay characteristic. Its main uses are listed below:\n\n- Reduce the prominence of quiet environmental noises (air conditioning, PC fans, clicks, etc.) during periods of silence\n- Suppress noise in quiet sections while maintaining the clarity of narration and dialogue\n- Emphasize the contours (punch) of sounds to create a more dynamic impression\n\n## Expander Options\n\nThe following is an explanation of each parameter when the mode is set to \"**Downward**.\"\n\n**Attack Time**: This determines the time it takes for the expander to begin suppressing the sound after the sound volume drops below the threshold. Shorter values will reduce the volume immediately, quickly suppressing noise and other imperfections. Longer values will cause the processing to begin later, preserving the natural reverberation and nuances of soft sounds.\n\n**Release Time**: This determines the time it takes for the expander to stop suppressing the volume and return to its original volume after the sound exceeds the threshold again and is determined to no longer require processing.\n\nLonger values will cause the sound to return more slowly, making it sound more natural.\n\nShorter values will return the sound to its original volume quickly, making it sound clearer and tighter, but may sound unnatural in some cases.\n\n**Attack Threshold**: The effect is only applied to sounds quieter than this value.\n\n**Release Threshold**: This can be the same as the Attack Threshold, but setting it about -5dB lower than the Attack Threshold will enable the Hysteresis function, suppressing fluttering around the threshold and achieving more stable processing.\n\n**Ratio**: This is an indicator of how much quieter (= how much) sounds below the threshold will be reduced. Set the value using the following as a guide:\n\n- A setting of 1 passes the signal as is (same as Expander OFF)\n- For light noise suppression (natural processing), use 1.5-3\n- For clarity of dialogue and smoothing out quiet sections, use 4-8\n- For complete noise reduction (gate-like behavior), use 20-100\n\n*Please note that setting the ratio too high can result in a choppy, unnatural sound.\n\n**Knee**: This parameter adjusts how smoothly processing begins near the threshold. The name comes from the fact that the curve on the graph resembles a knee. See below for the relationship between the setting value and the curve.\n\nKnee Value Volume Curve Characteristics Processing Sensation\n\n0dB Abrupt bend (hard knee) The effect is applied suddenly (like an ON/OFF switch)\n\n-6dB Gradual bend Processing begins naturally\n\n-12dB and above Very smooth Processing is barely noticeable\n\n**Makeup**: The gain applied after gating.\n\n**Dry Level**: The amount of unprocessed signal mixed with the output.\n\n**Wet Level**: The amount of processed signal mixed with the output.\n\nFor information on how to set up sidechaining and sidechain filtering, see the chapter of Gate.\n\n*When \"**Upward**\" is selected in the mode setting, the expander will function to boost (increase the volume of) quieter sounds.\n\nThis can be used in situations where dialogue or narration is too quiet to be heard, and is also effective when you want to bring out the subtle nuances of a musical performance (such as pianissimo).\n\nRather than \"lowering louder sounds\" like a compressor, it is used to increase the average volume by \"boosting quieter sounds,\" or to increase sound pressure while maintaining the punch and three-dimensionality of the original sound.\n\n\n# Filter\n\nA Filter is used to amplify (boost), pass or attenuate (cut) defined parts of a frequency spectrum. Easy Effects uses the Filter from Linux Studio Plugins.\n\n**Type** \n- **Low-pass** - Low-pass filter with rejection of high frequencies. \n- **High-pass** - High-pass filter with rejection of low frequencies. \n- **Low-shelf** - Shelving filter with adjustment of low frequencies. \n- **High-shelf** - Shelving filter with adjustment of high frequency range. \n- **Bell** - Bell filter with smooth peak/recess. \n- **Bandpass** - Bandpass filter. \n- **Notch** - Notch filter with full rejection of selected frequency. \n- **Resonance** - Resonance filter with sharp peak/recess. \n- **Ladder-pass** - The filter that makes some ladder-passing in the spectrum domain. \n- **Ladder-rej** - The filter that makes some ladder-rejection in the spectrum domain. \n- **Allpass** - All-pass filter which only affects the phase of the audio signal at the specified frequency. \n\n**Filter Mode** \n- **RLC** - Very smooth filters based on similar cascades of RLC contours. \n- **BWC** - Butterworth-Chebyshev-type-1 based filters. Does not affect Resonance and Notch filters. \n- **LRX** - Linkwitz-Riley based filters. Does not affect Resonance and Notch filters. \n- **APO** - Digital biquad filters derived from canonic analog biquad prototypes digitalized through Bilinear transform. These are textbook filters which are implemented as in the EqualizerAPO software. \n- **BT** - Bilinear Z-transform is used for pole/zero mapping. \n- **MT** - Matched Z-transform is used for pole/zero mapping. \n- **DR** - Direct design is used to serve the digital filter coefficients directly in the digital domain, without performing transforms. \n\n**Equalizer Mode** \n- **IIR** - Infinite Impulse Response filters, nonlinear minimal phase. In most cases does not add noticeable latency to output signal. \n- **FIR** - Finite Impulse Response filters with linear phase, finite approximation of equalizer's impulse response. Adds noticeable latency to output signal. \n- **FFT** - Fast Fourier Transform approximation of the frequency chart, linear phase. Adds noticeable latency to output signal. \n- **SPM** - Spectral Processor Mode of equalizer, equalizer transforms the magnitude of signal spectrum instead of applying impulse response to the signal.\n\n**Slope** \nThe slope of the filter characteristics. \n\n**Frequency** \nThe cutoff/resonance frequency of the filter or the middle frequency of the band. \n\n**Width** \nThe width of the bandpass/ladder filters in octaves. \n\n**Gain** \nThe gain of the filter. It is disabled for lo-pass/hi-pass/notch filters. \n\n**Quality** \nThe quality factor of the filter. \n\n**Balance** \nThe balance between left and right output channels. \n\n## References\n\n- [Wikipedia Audio Filter](https://en.wikipedia.org/wiki/Audio_filter)\n- [Linux Studio Plugins Filter](https://lsp-plug.in/?page=manuals§ion=filter_stereo)\n\n\n# Gate\n\nThe Gate attenuates signals that register below a Threshold. This kind of signal processing is used to reduce disturbing noise between useful signals. Easy Effects uses the Stereo Sidechain Gate from Linux Studio Plugins.\n\n## Gate Workflow\n\nThe Gate begins to open when the Sidechain level becomes above the Attack Zone Start level.\n\nThe Gate fully opens when the Sidechain level becomes above the Attack Threshold level.\n\nThe Gate begins to close when the Sidechain level becomes below the Release Zone Start level.\n\nThe Gate fully closes when the Sidechain level becomes below the Release Threshold level.\n\n## Gate Options\n\n**Attack Time** \nThe length of time it takes to restore roughly two-thirds of the gain reduction.\n\n**Release Time** \nThe length of time it takes to apply roughly two-thirds of the gain reduction.\n\n**Curve Threshold** \nThe Gate fully opens upon the Sidechain level becoming above Curve Threshold (displayed as Attack Threshold level).\n\nIf Hysteresis is not enabled, the Gate begins to close upon the Sidechain level becoming below Curve Threshold (displayed as Release Zone Start level).\n\n**Curve Zone Size** \nThe Gate begins to open upon the Sidechain level becoming above the Curve Threshold + Curve Zone (displayed as Attack Zone Start level).\n\nIf Hysteresis is not enabled, the Gate fully closes upon the Sidechain level becoming below the Curve Threshold + Curve Zone (displayed as Release Threshold level).\n\n**Hysteresis** \nWhen enabled, Curve Threshold and Curve Zone apply only to the opening Gate, and separate parameters can be configured for closing Gate.\n\n**Hysteresis Threshold** \nIf Hysteresis is enabled, the Gate begins to close upon the Sidechain level becoming below Curve Threshold + Hysteresis Threshold (displayed as Release Zone Start level).\n\n**Hysteresis Zone Size** \nIf Hysteresis is enabled, the Gate fully closes upon the Sidechain level becoming below the Curve Threshold + Hysteresis Threshold + Hysteresis Zone (displayed as Release Threshold level).\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal mixed in the output.\n\n**Reduction** \nIf the value is negative, it acts as the amount of gain reduction to apply to the input signal when the Gate is fully closed. If the value is positive, the Gate operates in \"Reverse Mode\": It reduces the amplitude when the signal is above the threshold.\n\n**Makeup** \nThe gain to apply after the gating stage.\n\n## Sidechain\n\n**Listen** \nAllows to listen the processed Sidechain signal.\n\n**Input Type** \nDetermines which signal is the Sidechain or, in other words, the signal that controls the gating stage.\n\n- **Internal** - The Sidechain is the Gate input signal (taken after applying the plugin input gain).\n- **External** - The Sidechain is an external source took by a specific input device (typically a microphone).\n\n**Input Device** \nSelect the device for the External Sidechain.\n\n**Mode** \nDetermines how the Sidechain is evaluated for the gating stage.\n\n- **Peak** - The Gate reacts according to the peaks.\n- **RMS** - The Gate reacts according to the average loudness measured by the root mean square.\n- **Low-Pass Filter** - The Gate reacts according to the signal processed by a recursive 1-pole Low-Pass filter.\n- **Simple Moving Average** - The Gate reacts according to the signal processed by the Simple Moving Average filter.\n\n**Source** \nDetermines which part of the Sidechain is taken into account for the gating stage.\n\n- **Middle** - The sum of left and right channels.\n- **Side** - The difference between left and right channels.\n- **Left** - Only left channel is used.\n- **Right** - Only right channel is used.\n- **Min** - The absolute minimum value is taken from stereo input.\n- **Max** - The absolute maximum value is taken from stereo input.\n\n**PreAmplification** \nGain applied to the Sidechain signal.\n\n**Reactivity** \nThe time that defines the number of samples used to process the Sidechain in RMS, Uniform and Low-Pass modes. Higher the value, more smooth the gating.\n\n**Lookahead** \nThe signal to gate is delayed by this amount of time, so that the gating will be applied earlier than it would be otherwise. The corresponding delay is reproduced on the output signal.\n\n## Sidechain Filters\n\n**High-Pass Filter Mode** \nSets the type of the High-Pass filter applied to Sidechain signal.\n\n**High-Pass Frequency** \nSets the cut-off frequency of the High-Pass filter.\n\n**Low-Pass Filter Mode** \nSets the type of the Low-Pass filter applied to Sidechain signal.\n\n**Low-Pass Frequency** \nSets the cut-off frequency of the Low-Pass filter.\n\n## References\n\n- [Wikipedia Dynamic Range Compression](https://en.wikipedia.org/wiki/Dynamic_range_compression)\n- [LSP Sidechain Gate Stereo](https://lsp-plug.in/?page=manuals§ion=sc_gate_stereo)\n- [Wikipedia Noise Gate](https://en.wikipedia.org/wiki/Noise_gate)\n\n\n# Harmonizer\n\nThe Harmonizer adds two pitch shifted voices to the input. Set the root note and chord, then choose an interval and level for each voice. The dry/wet control balances the original signal with the harmonized signal.\n\nEasy Effects uses the StereoHarm LV2 plugin from Rakarrack Plus. Install Rakarrack Plus to use this effect.\n\n## References\n\n- [Rakarrack Plus](https://github.com/Stazed/rakarrack-plus)\n- [Harmonizer documentation](https://github.com/Stazed/rakarrack-plus)\n\n\n# Limiter\n\nA Limiter is a special type of downward [Compressor](compressor.md) which does not allow the signal to overtake a predetermined Threshold. Ideally it has a very high compression ratio that takes the amplitude below a ceiling which stands as the maximum output level. For this reason it is usually named \"brick-wall limiter\".\n\nEasy Effects uses the Sidechain Stereo Limiter from Linux Studio Plugins. In most cases it works as a brick-wall limiter, but it offers also an additional feature that acts like a Compressor with extreme settings, so the output signal may exceed the specified Threshold.\n\n**Mode** \nSelect the operative mode of the peak cutting algorithm which searches the peaks above the Threshold and applies short gain reduction patches to the signal.\n\nThese patches can be selected in 3 forms: **Hermite**, **Exponential** and **Linear**. Each one has 4 different variants related to gain reduction of the samples around the peak: **Thin**, **Tail**, **Duck** and **Wide**.\n\nThe shape of each form and variant can be referred into the Linux Studio Plugin manual. See References section at the bottom of the present document.\n\n**Oversampling** \nWhen enabled, the sample rate is internally increased in order to improve peak detection and reduce aliasing (that causes distortion).\n\nThe modes have 2 main types: **Full**, which increases both the Sidechain and the Input signals, and **Half** which increases only the Sidechain. Each one of them specifies different **multipliers** and, between parentheses, the number of \"**lobes** in the kernel\".\n\nThe oversampled Input signal is downsampled to the original sample rate after processing.\n\n**Dither** \nIf specified, it enables Dithering for the selected bit depth. The process of Dithering adds a low-level noise to output signal in order to mask \"quantization distortion\", a form of artifact present in digital audio rendered at lower bit depth.\n\n**SC PreAmp** \nThe gain applied to the Sidechain before it is processed.\n\n**Lookahead** \nThe size of the buffer used to detect the peaks in advance. It adds the corresponding latency to the output signal.\n\n**Attack** \nThe length of time it takes to apply the needed gain reduction to keep the peak below the Threshold.\n\nIt affects the length of the gain reduction patch. It cannot be larger than the Lookahead (if specified larger, it's set internally as the max possible value).\n\n**Release** \nThe length of time it takes to restore the reduced gain around the limited peak.\n\nIt affects the length of the gain reduction patch. It cannot be twice larger than the Lookahead (if specified larger, it's set internally as the max possible value).\n\n**Threshold** \nThe target level above which the Limiter should reduce the peak of the signal. In some modes it represents the maximum output level.\n\n**Threshold Boost** \nIf enabled it applies an amount of gain to the limited signal equal to the absolute value of the Threshold. This causes the peak limited at the Threshold level to output at 0 dB (it has only effect when the Threshold is set below 0 dB).\n\n**Stereo Link** \nThe degree of the channel linking. At 0% both channels are limited independently while at 100% the loudest one triggers the same gain reduction on both.\n\n**External Sidechain** \nSwitch the Sidechain to an external source took by a specific input device (typically a microphone).\n\n## Auto Leveling\n\nThe Auto Leveling checkbutton introduces an additional feature named \"Automated Level Regulation\" (ALR) which acts like a Compressor with infinite ratio for the purpose of applying a smoothed gain reduction rather than a stronger peak cutter like in brick-wall mode. This configuration could get the output level to exceed the Threshold even if the signal is highly compressed.\n\n**Attack** \nManage how the raise of the input signal affects the smoothness of the ALR curve that controls the gain reduction level. Higher the value, more quickly the curve goes to it's maximum.\n\n**Release** \nManage how the fall of the input signal affects the smoothness of the curve that controls the gain reduction level. Higher the value, more quickly the curve goes to it's minimum.\n\n**Knee** \nManage the Threshold of the ALR gain curve and, in fact, adjust the balance between two gain reduction stages. Raising the value delegates more work to the peak-cutting algorithm. Lowering the value delegates more work to the ALR gain reduction algorithm.\n\n## References\n\n- [Wikipedia Limiter](https://en.wikipedia.org/wiki/Limiter)\n- [LSP Sidechain Limiter Stereo](https://lsp-plug.in/?page=manuals§ion=sc_limiter_stereo)\n\n\n# Loudness\n\nEasy Effects uses the Loudness Compensator from Linux Studio Plugins which applies the \"equal-loudness contour\" corrections to the input signal.\n\nAn equal-loudness contour is a measure of sound pressure level (SPL), over the frequency spectrum, for which a listener perceives a constant loudness when presented with pure steady tones. The unit of measurement for loudness levels is the phon and is arrived at by reference to equal-loudness contours.\n\nUsage of equal-loudness contours solves many mixing problems that every sound engineer meets while working on the track. The main problem is that human ear perceives different frequencies for different volume settings in a different way. In other words, applying changes to the mix on the low volume settings may cause unexpected sounding of the mix at the maximum loudness.\n\nThe Loudness Compensator performs frequency response computations and applies the computed frequency response to the input signal depending on the output volume settings. Additionally it can provide ear protection by applying hard-clipping to the output signal if it exceeds the allowed configurable level.\n\n**Standard** \nAllows to select different equal-loudness contour standards.\n\n- **Flat** - Applies flat frequency response to the whole spectrum. It's similar to just a gain knob but useful to perform a comparison to other modes.\n- **ISO 226:2003** - Recent equal-loudness contour standard published in 2003.\n- **Fletcher-Munson** - The first equal-loudness contour implementation by Harvey Fletcher and Wilden A. Munson published in 1933.\n- **Robinson-Dadson** - More accurate equal-loudness contour implementation by D.W. Robinson and R.S. Dadson published in 1956. It became the basis for the ISO 226:2003 standard.\n\n**FFT Size** \nAllows to select size of the Fast Fourier Transform frame used for the processing. The larger FFT frame is, the more precise the curve is approximated and the more latency is introduced.\n\n**Output Volume** \nThe output volume of the signal with applied equal loudness contour. It controls the loudness of the 1 kHz pure sine wave.\n\n**Clipping** \nAllows to enable and the hard clipping of the output signal.\n\n**Clipping Range** \nAllows to set the gap level for the hard clipping of the output signal.\n\n## References\n\n- [Wikipedia Equal-Loudness Contour](https://en.wikipedia.org/wiki/Equal-loudness_contour)\n- [LSP Loudness Compensator Stereo](https://lsp-plug.in/?page=manuals§ion=loud_comp_stereo)\n- [Lindos Electronics - Equal-Loudness Contours](http://www.lindos.co.uk/cgi-bin/FlexiData.cgi?SOURCE=Articles&VIEW=full&id=17)\n\n\n# Maximizer\n\nThe Maximizer is a special type of [Limiter](limiter.md) that does not only prevent the signal to exceed a specified target level, but also adjusts the average Loudness of the audio track.\n\nEasy Effects uses the Maximizer developed by ZamAudio. It acts like an Amplifier that feeds a brick-wall Limiter with a fixed Lookahead of 10 ms adding the corresponding delay to the output signal.\n\n**Threshold** \nThis parameter represents the ideal amplification level needed by the signal.\n\nIt contributes along with the Ceiling to determine the amount of gain to apply to the signal before the limiting stage. The gain is calculated by `(-Threshold) - (-Ceiling)`.\n\nFor example, on Threshold -6 dB and Ceiling -2 dB, the signal is amplified by 4 dB and limited to not exceed -2 dB output level.\n\n**Ceiling** \nThis parameter represents the ideal attenuation level needed by the signal and the maximum allowed output level.\n\nWhen the Threshold is set to 0 dB, the Ceiling is simply the gain reduction. When the Threshold is lowered, the signal is boosted without overtaking the Ceiling value.\n\n**Release** \nSets the release of the internal brick-wall Limiter. Lower values may introduce small artifacts.\n\n## References\n\n- [Sonarworks Blog - What Is A Maximizer?](https://www.sonarworks.com/soundid-reference/blog/learn/what-is-a-maximizer/)\n\n\n# Mid-Side Equalizer\n\nThe Mid-Side Equalizer processes the Mid (sum) and Side (difference) parts of a stereo signal independently. It's useful, for example, to fix the overall tone of a recording through the Mid channel while keeping the stereo ambience untouched on the Side channel. Easy Effects uses the Parametric Equalizer x32 MidSide from Linux Studio Plugins. You can run anywhere from 1 to 32 bands per channel, and tweak the width and center frequency of each band.\n\n## Global Options\n\n**Bands** \nNumber of bands.\n\n**Mode** \n- **IIR** - Infinite Impulse Response filters, nonlinear minimal phase. Usually doesn't add noticeable latency to the output.\n- **FIR** - Finite Impulse Response filters with linear phase, a finite approximation of the equalizer's impulse response. Adds noticeable latency to the output.\n- **FFT** - Fast Fourier Transform approximation of the frequency chart, linear phase. Adds noticeable latency to the output.\n\n**Balance** \nBalance between the left and right output channels.\n\n**Pitch Mid** \nFrequency shift for all mid channel filters, in semitones.\n\n**Pitch Side** \nFrequency shift for all side channel filters, in semitones.\n\n**Link Mid/Side** \nWhen enabled, you can apply different configurations to the mid and side channels.\n\n**Flat Response** \nResets all band gains to 0.\n\n**Calculate Frequencies** \nFigures out the center frequency and width for each band based on the current band count. Handy if you want fewer than 32 bands but aren't sure which frequencies to pick.\n\n## Band Options\n\n**Type** \n- **Off** - Filter is disabled.\n- **Bell** - Bell filter with smooth peak/recess.\n- **High Pass** - High Pass filter that removes low frequencies.\n- **High Shelf** - Shelving filter that adjusts the high frequency range.\n- **Low Pass** - Low Pass filter that removes high frequencies.\n- **Low Shelf** - Shelving filter that adjusts the low frequency range.\n- **Notch** - Notch filter that fully rejects the selected frequency.\n- **Resonance** - Resonance filter with sharp peak/recess.\n- **All Pass** - All Pass filter.\n\n**Mode** \n- **RLC** - Very smooth filters built from cascaded RLC contours. Uses Bilinear Z-transform (BT) or Matched Z-transform (MT) for pole/zero mapping.\n- **BWC** - Butterworth-Chebyshev-type-1 based filters. Doesn't affect Resonance and Notch filters. Uses Bilinear Z-transform (BT) or Matched Z-transform (MT) for pole/zero mapping.\n- **LRX** - Linkwitz-Riley based filters. Doesn't affect Resonance and Notch filters. Uses Bilinear Z-transform (BT) or Matched Z-transform (MT) for pole/zero mapping.\n- **APO** - Digital biquad filters derived from canonical analog biquad prototypes digitalized through Bilinear transform. These are [textbook filters](https://shepazu.github.io/Audio-EQ-Cookbook/audio-eq-cookbook.html) implemented the same way as in [Equalizer APO](https://equalizerapo.com/). Direct design (DR) generates the digital filter coefficients directly in the digital domain, without any transforms.\n\n**Slope** \nSlope of the filter curve.\n\n**Solo** \nMakes the selected band the only active one.\n\n**Mute** \nMutes the selected band.\n\n**Frequency** \nCenter frequency of the band.\n\n**Width** \nBandwidth, calculated as `width = frequency / quality`.\n\n**Quality** \nQuality factor of the filter.\n\n## References\n\n- [Wikipedia Equalization (audio)](https://en.wikipedia.org/wiki/Equalization_(audio))\n- [LSP Parametric Equalizer x32 MidSide](http://lsp-plug.in/?page=manuals§ion=para_equalizer_x32_ms)\n- [Wikipedia Q Factor](https://en.wikipedia.org/wiki/Q_factor)\n- [How to EQ - Q Factor and Bandwidth in EQ: What They Mean](https://howtoeq.wordpress.com/2010/10/07/q-factor-and-bandwidth-in-eq-what-it-all-means/)\n\n\n# Multiband Compressor\n\nEasy Effects uses the Sidechain Multiband Compressor Stereo developed by Linux Studio Plugins. Please refer to the [Compressor](compressor.md) documentation to comprehend the basic functionality of the dynamic range compression.\n\n## Global Options\n\n**Band Management** \nThis Compressor allows to split the input signal up to 8 bands. Each band is not attached to its strict frequency and can be controlled by completely different frequency range that can be obtained by applying Low-Cut and Hi-Cut filters to the Sidechain signal.\n\nThe first band is always enabled while the others can be activated if needed. When only the first band is enabled, the functionality is similar to the Singleband Sidechain Compressor. Otherwise the signal is split in more bands and each band is compressed individually. After the compression stage, the bands are mixed together to form the output result.\n\n**Operating Mode** \nDetermines how the input signal is split to obtain the different bands.\n\n- **Classic** - The original signal is split using crossover filters. After the compression stage, all bands become phase-compensated using All-Pass filters.\n- **Modern** - Each band is processed by a pair of dynamic shelving filters. This mode allows a better control over the gain of each band.\n\n**Sidechain Boost** \nIf enabled, it introduces a special mode for assigning the same weight for higher frequencies opposite to lower frequencies.\n\nWhen disabled, the frequency band is processed 'as is', but the usual audio signal has 3 dB/octave falloff in the frequency domain and could be compared with the pink noise. So lower frequencies take more effect on the Compressor rather than higher frequencies.\n\nOn the other hand the Sidechain Boost mode allows to compensate the -3 dB/octave falloff of the signal spectrum and, even more, makes the signal spectrum growing +3 dB/octave in the almost fully audible frequency range.\n\n**Pink** applies +3 dB/octave while **Brown** applies +6 dB/octave Sidechain Boost. Each of them can use bilinear-transformed (BT) or matched-transformed (MT) shelving filter.\n\n**Sidechain Source** \nIf the External Sidechain is enabled inside at least one band, this combobox allows to select the input device as source.\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal mixed in the output.\n\n## Band Options\n\n**Band Start** \nAllows to change the lower end split frequency of the selected band. This value is assigned to the Band End of the previous enabled band. It can be modified for all bands except the first one, which is always enabled and its value is 0 Hz.\n\n**Band End** \nSpecify the upper end split frequency of the selected band. It cannot be directly modified and assumes the same value of the Band Start of the next enabled band. For the last enabled band it is always 24.000 Hz.\n\n**Compression Mode** \n- **Downward** - It's aimed to decrease the gain of the signal above the Threshold.\n- **Upward** - It's aimed to increase the gain of the signal below the Threshold.\n- **Boosting** - It's aimed to increase the gain of the signal below the Threshold by a specific amount.\n\n**External Sidechain** \nThe Sidechain is an external source took by a specific input device (typically a microphone).\n\n**Band Bypass** \nIf enabled, the selected band is not affected by the compression stage.\n\n**Solo** \nTurns on the Solo mode to the selected band by applying -36 dB gain to the other non-soloing bands.\n\n**Mute** \nTurns on the Mute mode applying -36 dB gain to to the selected band.\n\n**Attack Time** \nThe length of time it takes to apply roughly two-thirds of the targeted amount of compression ratio to the uncompressed band signal.\n\n**Release Time** \nThe length of time it takes to restore roughly two-thirds of the reduced gain (in Downward mode) or increased gain (in Upward/Boosting mode) to the compressed band signal.\n\n**Attack Threshold** \nThe target level around which the compression is applied (the range depends by the Knee).\n\n**Release Threshold** \nSets up the Threshold of the Release Time, calculated by summing the Release Threshold to the Attack Threshold.\n\nIf the Sidechain level is above that Threshold, the compressor uses the Release Time for the releasing stage. Otherwise the Attack Time is used in place of Release Time.\n\nFor example, with -10 dB Attack Threshold and -60 dB Release Threshold, if the Sidechain is above `-10 + (-60) = -70 dB`, the Release Time is used for the gain restoration. If the Sidechain is below -70 dB, the Attack Time is used for the gain restoration.\n\n**Ratio** \nThe amount of attenuation (in Downward mode) or amplification (in Upward/Boosting mode) that will be applied to the signal.\n\nFor example, when the Ratio is 2 in Downward mode and the Sidechain rises above the Threshold by 10 dB, the signal would be ideally reduced by 5 dB (10/2 dB). In practice this behavior mostly depends on how the Multiband Compressor is designed and configured.\n\n**Knee** \nThe range over which the Compressor switches from no compression to almost the full ratio compression (the Threshold usually sits at the center of this transition zone).\n\n**Makeup** \nThe gain to apply after the compression stage.\n\n## Band Sidechain Options\n\n**Mode** \nDetermines how the Sidechain of the selected band is evaluated for the compression stage.\n\n- **Peak** - The Compressor reacts according to the peaks.\n- **RMS** - The Compressor reacts according to the average loudness measured by the root mean square.\n- **Low-Pass** - The Compressor reacts according to the signal processed by a Low-Pass filter.\n- **Uniform** - The Compressor reacts according to the loudness measured by the average of the absolute amplitude.\n\n**Source** \nDetermines which part of the Sidechain is taken into account for the compression stage.\n\n- **Middle** - The sum of left and right channels.\n- **Side** - The difference between left and right channels.\n- **Left** - Only left channel is used.\n- **Right** - Only right channel is used.\n- **Min** - The absolute minimum value is taken from stereo input.\n- **Max** - The absolute maximum value is taken from stereo input.\n\n**Low-Cut Filter** \nEnables a custom Low-Cut Filter for the selected band.\n\n**Low-Cut Frequency** \nSets the cut-off frequency of the custom Low-Cut filter. If it is disabled, the default Low-Cut filter assumes internally the Band Start frequency as the cut-off frequency.\n\n**Hight-Cut Filter** \nEnables a custom High-Cut Filter for the selected band.\n\n**Hight-Cut Frequency** \nSets the cut-off frequency of the custom High-Cut filter. If it is disabled, the default High-Cut filter assumes internally the Band End frequency as the cut-off frequency.\n\n**PreAmp** \nGain applied to the Sidechain signal of the selected band.\n\n**Reactivity** \nThe time that defines the number of samples used to process the Sidechain in RMS, Uniform and Low-Pass modes. Higher the value, more smooth the compression.\n\n**Lookahead** \nThe band signal to compress is delayed by this amount of time, so that the compression will be applied earlier than it would be otherwise.\n\nEach band can have different Lookahead values. To avoid phase distortions in the mixing stage, all the bands are automatically delayed for an individually calculated period of time.\n\n**Boost Amount** \nMaximum gain amplification to apply in Boosting Mode.\n\n**Boost Threshold** \nThe Threshold below which a constant amplification will be applied to the band signal in Upward Mode (it prevents from applying infinite amplification to very quiet signals).\n\n## References\n\n- [Wikipedia Dynamic Range Compression](https://en.wikipedia.org/wiki/Dynamic_range_compression)\n- [LSP Sidechain Multiband Compressor Stereo](https://lsp-plug.in/?page=manuals§ion=sc_mb_compressor_stereo)\n- [Black Ghost Audio - The Ultimate Guide to Compression](https://www.blackghostaudio.com/blog/the-ultimate-guide-to-compression)\n- [Attack Magazine - Demolishing The Myths of Compression](https://www.attackmagazine.com/features/columns/gregory-scott-demolishing-the-myths-of-compression/)\n\n\n# Multiband Gate\n\nEasy Effects uses the Multiband Gate Stereo developed by Calf Studio Gear. Please refer to the [Gate](gate.md) documentation to comprehend the basic functionality of the gating process.\n\n## Global Options\n\n**Band Management** \nThis Gate allows to split the input signal up to 8 bands. Each band is not attached to its strict frequency and can be controlled by completely different frequency range that can be obtained by applying Low-Cut and Hi-Cut filters to the Sidechain signal.\n\nThe first band is always enabled while the others can be activated if needed. When only the first band is enabled, the functionality is similar to the Singleband Sidechain Gate. Otherwise the signal is split in more bands and each band is processed individually. After the gating stage, the bands are mixed together to form the output result.\n\n**Operating Mode** \nDetermines how the input signal is split to obtain the different bands.\n\n- **Classic** - The original signal is split using crossover filters. After the gating stage, all bands become phase-compensated using All-Pass filters.\n- **Modern** - Each band is processed by a pair of dynamic shelving filters. This mode allows a better control over the gain of each band.\n\n**Sidechain Boost** \nIf enabled, it introduces a special mode for assigning the same weight for higher frequencies opposite to lower frequencies.\n\nWhen disabled, the frequency band is processed 'as is', but the usual audio signal has 3 dB/octave falloff in the frequency domain and could be compared with the pink noise. So lower frequencies take more effect on the Gate rather than higher frequencies.\n\nOn the other hand the Sidechain Boost mode allows to compensate the -3 dB/octave falloff of the signal spectrum and, even more, makes the signal spectrum growing +3 dB/octave in the almost fully audible frequency range.\n\n**Pink** applies +3 dB/octave while **Brown** applies +6 dB/octave Sidechain Boost. Each of them can use bilinear-transformed (BT) or matched-transformed (MT) shelving filter.\n\n**Sidechain Source** \nIf the External Sidechain is enabled inside at least one band, this combobox allows to select the input device as source.\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal mixed in the output.\n\n## Band Options\n\n**Band Start** \nAllows to change the lower end split frequency of the selected band. This value is assigned to the Band End of the previous enabled band. It can be modified for all bands except the first one, which is always enabled and its value is 0 Hz.\n\n**Band End** \nSpecify the upper end split frequency of the selected band. It cannot be directly modified and assumes the same value of the Band Start of the next enabled band. For the last enabled band it is always 24.000 Hz.\n\n**External Sidechain** \nThe Sidechain is an external source took by a specific input device (typically a microphone).\n\n**Band Bypass** \nIf enabled, the selected band is not affected by the gating stage.\n\n**Solo** \nTurns on the Solo mode to the selected band by applying -36 dB gain to the other non-soloing bands.\n\n**Mute** \nTurns on the Mute mode applying -36 dB gain to to the selected band.\n\n**Attack Time** \nThe length of time it takes to apply roughly two-thirds of the gain reduction.\n\n**Release Time** \nThe length of time it takes to restore roughly two-thirds of the gain reduction.\n\n**Curve Threshold** \nThe Gate fully opens upon the Sidechain level becoming above Curve Threshold.\n\nIf Hysteresis is not enabled, the Gate begins to close upon the Sidechain level becoming below Curve Threshold.\n\n**Curve Zone Size** \nThe Gate begins to open upon the Sidechain level becoming above the Curve Threshold + Curve Zone.\n\nIf Hysteresis is not enabled, the Gate fully closes upon the Sidechain level becoming below the Curve Threshold + Curve Zone.\n\n**Hysteresis** \nWhen enabled, Curve Threshold and Curve Zone apply only to the opening Gate, and separate parameters can be configured for closing Gate.\n\n**Hysteresis Threshold** \nIf Hysteresis is enabled, the Gate begins to close upon the Sidechain level becoming below Curve Threshold + Hysteresis Threshold.\n\n**Hysteresis Zone Size** \nIf Hysteresis is enabled, the Gate fully closes upon the Sidechain level becoming below the Curve Threshold + Hysteresis Threshold + Hysteresis Zone.\n\n**Reduction** \nIf the value is negative, it acts as the amount of gain reduction to apply to the input signal when the Gate is fully closed. If the value is positive, the Gate operates in \"Reverse Mode\": It reduces the amplitude when the signal is above the threshold.\n\n**Makeup** \nThe gain to apply after the gating stage.\n\n## Band Sidechain Options\n\n**Mode** \nDetermines how the Sidechain is evaluated for the gating stage.\n\n- **Peak** - The Gate reacts according to the peaks.\n- **RMS** - The Gate reacts according to the average loudness measured by the root mean square.\n- **Low-Pass Filter** - The Gate reacts according to the signal processed by a recursive 1-pole Low-Pass filter.\n- **Simple Moving Average** - The Gate reacts according to the signal processed by the Simple Moving Average filter.\n\n**Source** \nDetermines which part of the Sidechain is taken into account for the gating stage.\n\n- **Middle** - The sum of left and right channels.\n- **Side** - The difference between left and right channels.\n- **Left** - Only left channel is used.\n- **Right** - Only right channel is used.\n- **Min** - The absolute minimum value is taken from stereo input.\n- **Max** - The absolute maximum value is taken from stereo input.\n\n**Low-Cut Filter** \nEnables a custom Low-Cut Filter for the selected band.\n\n**Low-Cut Frequency** \nSets the cut-off frequency of the custom Low-Cut filter. If it is disabled, the default Low-Cut filter assumes internally the Band Start frequency as the cut-off frequency.\n\n**Hight-Cut Filter** \nEnables a custom High-Cut Filter for the selected band.\n\n**Hight-Cut Frequency** \nSets the cut-off frequency of the custom High-Cut filter. If it is disabled, the default High-Cut filter assumes internally the Band End frequency as the cut-off frequency.\n\n**PreAmp** \nGain applied to the Sidechain signal of the selected band.\n\n**Reactivity** \nThe time that defines the number of samples used to process the Sidechain in RMS, Uniform and Low-Pass modes. Higher the value, more smooth the gating.\n\n**Lookahead** \nThe band signal to gate is delayed by this amount of time, so that the gating will be applied earlier than it would be otherwise.\n\nEach band can have different Lookahead values. To avoid phase distortions in the mixing stage, all the bands are automatically delayed for an individually calculated period of time.\n\n## References\n\n- [LSP Sidechain Multiband Gate Stereo](https://lsp-plug.in/?page=manuals§ion=sc_mb_gate_stereo)\n- [Wikipedia Noise Gate](https://en.wikipedia.org/wiki/Noise_gate)\n\n\n# Pitch\n\nPitch shifting is a sound recording technique in which the original Pitch of a sound is raised or lowered. Easy Effects uses the pitch shifter from SoundTouch.\n\n**Mode** \nControls the method used for pitch shifting.\n\n- **High Speed** - Uses a method with a CPU cost that is relatively moderate and predictable.\n- **High Quality** - Uses the highest quality method for pitch shifting. This CPU cost is approximately proportional to the required frequency shift.\n- **High Consistency** - Uses the method that gives greatest consistency when used to create small variations in pitch around the 1.0-ratio level. Unlike the previous two options, this avoids discontinuities when moving across the 1.0 pitch scale. It also consumes more CPU than the others in the case where the pitch scale is exactly 1.0.\n\n**Formant** \nControls the handling of formant shape (spectral envelope) when pitch-shifting.\n\n- **Shifted** - Applies no special formant processing. The spectral envelope will be pitch shifted as normal.\n- **Preserved** - Preserves the spectral envelope of the unshifted signal. This permits shifting the note frequency without so substantially affecting the perceived pitch profile of the voice or instrument.\n\n**Transients** \nControls the component frequency phase-reset mechanism that may be used at transient points to provide clarity and realism to percussion and other significant transient sounds.\n\n- **Crisp** - Resets component phases at the peak of each transient (the start of a significant note or percussive event). This usually results in a clear-sounding output, but it is not always consistent, and may cause interruptions in stable sounds present at the same time as transient events.\n- **Mixed** - Resets component phases at the peak of each transient, outside a frequency range typical of musical fundamental frequencies. The results may be more regular for mixed stable and percussive notes than Crisp option, but with a \"phasier\" sound. The balance may sound very good for certain types of music and fairly bad for others.\n- **Smooth** - Does not reset component phases at any point. The results will be smoother and more regular but may be less clear than the other transient options.\n\n**Detector** \nControls the type of transient detector used.\n\n- **Compound** - Uses a general purpose transient detector which is likely to be good for most situations.\n- **Percussive** - Detects percussive transients.\n- **Soft** - Uses an onset detector with less of a bias toward percussive transients. This may give better results with certain material (e.g. relatively monophonic piano music).\n\n**Phase** \nControls the adjustment of component frequency phases from one analysis window to the next during non-transient segments.\n\n- **Laminar** - Adjusts phases when stretching in such a way as to try to retain the continuity of phase relationships between adjacent frequency bins whose phases are behaving in similar ways. This should give good results in most situations.\n- **Independent** - Adjusts the phase in each frequency bin independently from its neighbours. This usually results in a slightly softer, phasier sound.\n\n**Cents** \nNumber of cents the Pitch will be increased or decreased.\n\n**Semitones** \nNumber of semitones the Pitch will be increased or decreased.\n\n**Octaves** \nNumber of octaves the Pitch will be increased or decreased.\n\n## References\n\n- [Wikipedia Pitch Shift](https://en.wikipedia.org/wiki/Pitch_shift)\n- [SoundTouch Audio Time Stretcher Library - Attractive Features](https://www.surina.net/soundtouch/)\n\n\n# Reverberation\n\nReverberation is the phenomenon of persistence of sound after the source has been stopped as a result of multiple reflections of the waves over objects within a closed surface. These reflections build up with each other and decay gradually as they are absorbed by the surfaces of objects in the space enclosed.\n\nThe Reverberation is different than Echo because the Echo is a reflected sound wave with sufficient magnitude and delay to be detectable as a signal distinct from the source one. To simulate the Reverberation effect, Easy Effects uses the Reverb plugin developed by Calf Studio Gear.\n\n**High Frequency Damping** \nCutoff frequency of the reflections. It causes higher frequencies to decay faster.\n\n**Room Size** \nSize of the space where simulated Reverberation occurs inside. It determines the time between reflections.\n\n**Diffusion** \nDegree of uniformity. Higher values lead to less uniform Reverberation.\n\n**Pre Delay** \nAdditional delay. It corresponds to a distance between sound source and the nearest wall.\n\n**Decay Time** \nThe time it takes for Reverberation to fade out.\n\n**Dry Level** \nAmount of unprocessed signal mixed in the output.\n\n**Wet Level** \nAmount of processed signal (Reverberation) mixed in the output.\n\n**Bass Cut** \nRemoves low frequencies from the Reverberation.\n\n**Treble Cut** \nRemoves high frequencies from the Reverberation.\n\n## References\n\n- [Wikipedia Reverberation](https://en.wikipedia.org/wiki/Reverberation)\n- [Calf Reverb](https://calf-studio-gear.org/doc/Reverb.html)\n\n\n# Noise Reduction\n\nThe Noise Reduction is a process aimed to attenuate the disturbing noise from a signal.\n\nEasy Effects Noise Reduction is made on the RNNoise library which is based based on recurrent neural network, a class of artificial neural networks where connections between nodes form a directed graph along a temporal sequence. This allows it to exhibit temporal dynamic behavior.\n\nStandard RNNoise Model is used and custom models can be imported to perform different types of noise reduction.\n\n## References\n\n- [Wikipedia Noise Reduction](https://en.wikipedia.org/wiki/Noise_reduction)\n- [Wikipedia Recurrent Neural Network](https://en.wikipedia.org/wiki/Recurrent_neural_network)\n- [Jean-Marc Valin - RNNoise: Learning Noise Suppression](https://jmvalin.ca/demo/rnnoise/)\n\n\n# Spectrum\n\n**Enabled** \nShow or hide the Spectrum.\n\n**Shape** \nChoose the style of the animation.\n\n**Points** \nNumber of points/bars on the x axis.\n\n**Height** \nChanges the height of the Spectrum.\n\n**Line Width** \nThe width of a single bar.\n\n**Fill** \nDraw filled bars.\n\n**Show Bars Border** \nDraw bars with borders.\n\n**Rounded Corners** \nDraw bars with rounded corners.\n\n**Color** \nChoose the color of the Spectrum components.\n\n**Minimum Frequency Range** \nLower end frequency of the Spectrum.\n\n**Maximum Frequency Range** \nUpper end frequency of the Spectrum.\n\n[Settings Menu](../user_interface/settingsmenu.md)\n\n\n"}