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How to Recreate Classic Echoes and Delays with Ableton | Reverb News
The digital age set in—and you can hear it on so many songs that came out during this time. Single, smaller digital delay pedals gave more musicians access to delay effects than ever before. Soon, they were used on pretty much anything, from synths to vocals to guitars and beyond. Boss were the heavy-hitters in this era, with their DD-2 stompbox and eventually the DD-3 in We focused on making this feel like a Juno thrown into a DD-3 pedal and fed back in for record.
It is dry and to-the-point, without any modulations. To achieve these sounds, you can use a combination of long, feedback-heavy delays and reverbs. On a guitar, these effects together can create a wall of sound.
Reverse reverbs played a big role in this sound alongside the beloved Big Muff distortions and fuzz. When it comes to delays, its all about modulations and being able to pitch shift the delays to create bigger harmonies.
To achieve this tone, we sent the guitar into both the new Ableton 10 Pedal Plugin which emulates a guitar pedal and the Echo plugin with two instances of Echo—one to give an even more reverberated, short delay signal and one to handle the longer delay signal and echo duties. You can add your favorite reverb plugin and tweak as needed to build it even more. Our audio example actually sounds more modulated than it really is.
Thanks to software-based plugins and productions, the concepts were never quicker or easier to use. Many realized you could have more control, and, in return, that meant you can use the effects more thoughtfully, in more subtle ways. This made it common to apply different delay types to different moments of the song, as needed. This type of delay revolves around precise panning techniques and filters to showcase the flexibility found in a range of software plugins.
The Morph filter has an additional Morph control which sweeps the filter type continuously from lowpass to bandpass to highpass to notch and back to lowpass. You can adjust Frequency and Resonance by clicking and dragging in the X-Y controller or via the knobs. You can also click on the Freq and Res numeric displays and type in exact values. When using the non-Clean circuit types, the Resonance control allows for self-oscillation.
The pitch of the self-oscillation depends on both the Frequency and Resonance values. The Envelope section controls how the envelope modulation affects the filter frequency. The Amount control defines the extent to which the envelope affects the filter frequency, while the Attack control sets how the envelope responds to rising input signals.
Low Attack values cause a fast response to input levels; high values integrate any changes gradually, creating a looser, slower response. Think of it as adding inertia to the response. Lower Release values cause the envelope to respond more quickly to falling input signals.
Normally, the signal being filtered and the input source that triggers the envelope follower are the same signal.
But by using sidechaining , it is possible to filter a signal based on the envelope of another signal. To access the Sidechain parameters, unfold the Auto Filter window by toggling the button in its title bar. Enabling this section with the Sidechain button allows you to select another track from the choosers below.
Note that increasing the gain does not increase the volume of the source signal in the mix. The sidechain audio is only a trigger for the envelope follower and is never actually heard. The Auto Filter also contains a Low Frequency Oscillator to modulate filter frequency in a periodic fashion. The respective Amount control sets how much the LFO affects the filter.
This can be used in conjunction with or instead of the envelope follower. The Rate control specifies the LFO speed. It can be set in terms of hertz, or synced to the song tempo, allowing for controlled rhythmic filtering.
Available LFO waveform shapes are sine creates smooth modulations with rounded peaks and valleys , square, triangle, sawtooth up, sawtooth down, and sample and hold generates random positive and negative modulation values in mono and stereo. There are two LFOs, one for each stereo channel. Spin detunes the two LFO speeds relative to each other. Each stereo channel is modulated at a different frequency, as determined by the Spin amount. Instead, the Auto Filter offers two kinds of sample and hold: The upper sample and hold type available in the chooser provides independent random modulation generators for the left and right channels stereo , while the lower one modulates both channels with the same signal mono.
The Quantize Beat control applies quantized modulation to the filter frequency. With Quantize Beat off, frequency modulation follows the control source the Envelope, LFO, or manually-adjusted cutoff.
If you open a Set that was created in a version of Live older than version 9. These consist of 12 dB or 24 dB lowpass, bandpass and highpass filters, as well as a notch filter, and do not feature a Drive control.
Each Auto Filter loaded with the legacy filters shows an Upgrade button in the title bar. Pressing this button will permanently switch the filter selection to the newer models for that instance of Auto Filter.
Note that this change may make your Set sound different. Auto Pan offers LFO-driven manipulation of amplitude and panning for creating automatic panning, tremolo and amplitude modulation, and beat-synchronized chopping effects.
LFO speed is controlled with the Rate control, which can be set in terms of hertz. Rate can also be synced to the song tempo. Though both LFOs run at the same frequency, the Phase control lends the sound stereo movement by offsetting their waveforms relative to each other. Phase is particularly effective for creating vibrato effects. Beat Repeat allows for the creation of controlled or randomized repetitions of an incoming signal. The Interval control defines how often Beat Repeat captures new material and begins repeating it.
Gate defines the total length of all repetitions in sixteenth notes. Activating the Repeat button bypasses all of the above controls, immediately capturing material and repeating it until deactivated.
The Grid control defines the grid size — the size of each repeated slice. Large grid values create rhythmic loops, while small values create sonic artifacts.
The No Triplets button sets grid division as binary. Grid size can be changed randomly using the Variation control. But when Variation is set to higher values, the grid fluctuates considerably around the set Grid value. Pitch is adjusted through resampling in Beat Repeat, lengthening segments to pitch them down without again compressing them to adjust for the length change.
This means that the rhythmical structure can become quite ambiguous with higher Pitch values. The Pitch Decay control tapers the pitch curve, making each repeated slice play lower than the previous one. Warning: This is the most obscure parameter of Beat Repeat. Beat Repeat includes a combined lowpass and highpass filter for defining the passed frequency range of the device. You can turn the filter on and off, and set the center frequency and width of the passed frequency band, using the respective controls.
Gate mode is especially useful when the effect is housed in a return track. You can set the output level of the device using the Volume control, and apply Decay to create gradually fading repetitions.
Cabinet is an effect that emulates the sound of five classic guitar cabinets. Developed in collaboration with Softube, Cabinet uses physical modelling technology to provide a range of authentic sounds, with optimized mics and mic positioning. The Speaker chooser allows you to select from a variety of speaker sizes and combinations. In the real world, more and larger speakers generally means higher volumes. The Microphone chooser changes the position of the virtual microphone in relation to the speaker cabinet.
Near On-Axis micing results in a bright, focused sound, while Near Off-Axis is more resonant and a bit less bright. The switch below the Microphone chooser toggles between a Dynamic and Condenser mic. Dynamic mics are a bit grittier and commonly used when close-micing guitar cabinets because they are capable of handling much higher volumes. Condenser mics are more accurate, and are commonly used for micing from a distance. Guitar cabinets are normally fed by guitar amps.
For this reason, Cabinet is paired with Amp see But you can also achieve interesting and exotic sounds by using Amp and Cabinet separately. A common studio technique is to use multiple mics on a single cabinet, and then adjust the balance during mixing. Try this:. Inspired by EQs found on classic mixing desks, Channel EQ is a simple, yet flexible three-band EQ, fine-tuned to provide musical results for a variety of audio material. Activating the HP 80 Hz switch will toggle a high-pass filter, which is useful for removing the rumble from a signal.
The Low parameter controls the gain of a low shelf filter, tuned to Hz. The filter curve is adaptive and will change dynamically relative to the amount of gain applied. The Mid parameter controls the gain of a sweepable bell filter. The frequency slider located above the Mid control allows you to set the center frequency of this filter from Hz to 7.
When boosting, the High parameter controls the gain of a high shelf filter, up to 15 dB. When attenuating, the shelving filter is combined with a low-pass filter. Turning the parameter from 0 dB towards dB will simultaneously reduce the cutoff frequency of the low-pass filter from 20 kHz to 8 kHz.
A spectrum visualization provides real-time visual feedback of the resulting filter curves and processed signal. The Output control sets the amount of gain applied to the processed signal, and can be used to compensate for any changed signal amplitude resulting from the EQ settings. You can also shape the sound of a single drum or an entire drum kit, by placing an instance of Channel EQ onto one or multiple Drum Rack pads.
Adding an instance of Saturator see In such cases, boosting the low end considerably would also lead to increased distortion, similar to the behavior of analog mixing desks. In Chorus is no longer part of the Core Library as of Live Sets that were made in earlier versions, e. The Chorus effect uses two parallel time-modulated delays to create chorus thickening and flanging effects. Each delay has its own delay time control, calibrated in milliseconds.
Delay 1 has a highpass filter that can remove low frequencies from the delayed signal. Greater highpass values let only very high frequencies pass through to Delay 1. Delay 2 can switch among three different modes. When off, only Delay 1 is audible. When Mod is activated, Delay 2 will receive the same modulation as Delay 1.
This is especially useful if you want to change both delays with a single gesture. To change the modulation rate for the delay times, click and drag along the horizontal axis. To change the amount of modulation, click and drag along the vertical axis. You can also make changes by entering parameter values in the Amount and Rate fields below the X-Y controller. The Amount value is in milliseconds, while the modulation frequency rate is in Hertz. The Feedback control determines how much of the output signal feeds back into the input, while the Polarity switch sets surprise!
Polarity changes have the most effect with high amounts of feedback and short delay times. Set it to percent when using Chorus in a return track. This is enabled by default, except when loading Sets that use Chorus and were made in earlier versions of Live.
Chorus-Ensemble offers a classic two-delay line chorus with an optional third delay line mode. With a wide variety of tools for thickening sounds, creating flanging and vibrato effects, this device also allows you to easily recreate string ensemble chorus sounds.
Three different modes are provided, which can be chosen in the display: Classic, Ensemble, and Vibrato. Classic mode creates a thickened sound by adding two time-modulated delayed signals to the input. Use it for a classic chorus sound, adding light motion to your audio signal. Ensemble mode is based on and shares controls with Classic mode, but creates richer, smoother, and more intense chorus sound by using three delayed signals with evenly split modulation phase offsets.
Vibrato mode applies stronger modulation than a chorus to create pitch variation. In addition to the mode selector buttons, the display also provides access to a high-pass filter and the Width parameter. Width is active in Classic and Ensemble modes, but while in Vibrato mode, this parameter is replaced by Offset and Shape controls.
When enabled, the high-pass filter reduces the chorus effect on signal components below the frequency set by the High-Pass Frequency slider, ranging from 20 Hz to Hz. Width sets the stereo width of the wet signal, which in turn adjusts the chorus level balance between the mid and side channels.
This is used for maintaining the level of the effect across the stereo field, which can be helpful during mixing. When using Vibrato mode, Offset adjusts the amount of phase offset between the waveforms for the left and right channel. Shape enables you to change the shape of the modulation waveform between a sine and a triangle.
Rate sets the modulation rate in Hertz, and can be adjusted either with the dial or by dragging up or down in the display. Tip: Turn up Rate for a more drastic chorus sound, or keep it low for more gentle phasing.
Amount adjusts the amplitude of the modulation signals that affects delay times. Higher values result in a stronger time deviation from the unmodulated time setting. Increasing this sounds more extreme and tends to increase upper harmonic material, and will also create audible delays if playback is stopped.
Note that this is disabled in Vibrato mode. A compressor reduces gain for signals above a user-settable threshold. Compression reduces the levels of peaks, opening up more headroom and allowing the overall signal level to be turned up. The Threshold slider sets where compression begins.
Signals above the threshold are attenuated by an amount specified by the Ratio parameter, which sets the ratio between the input and output signal. For example, with a compression ratio of 3, if a signal above the threshold increases by 3 dB, the compressor output will increase by only 1 dB.
If a signal above the threshold increases by 6 dB, then the output will increase by only 2 dB. A ratio of 1 means no compression, regardless of the threshold. The orange Gain Reduction meter shows how much the gain is being reduced at any given moment. The more reduction, the more audible the effect; a gain reduction above 6 dB or so might produce the desired loudness, but significantly alters the sound and is easily capable of destroying its dynamic structure. This is something that cannot be undone in later production steps.
Keep this in mind especially when using a compressor, limiter or sound loudness-maximizing tool in the master channel. Less is often more here. Because compression reduces the volume of loud signals and opens up headroom, you can use the Output Out control so that the peaks once again hit the maximum available headroom.
Enabling the Makeup button automatically compensates the output level if the threshold and ratio settings change. The Knee control adjusts how gradually or abruptly compression occurs as the threshold is approached.
With a setting of 0 dB, no compression is applied to signals below the threshold, and full compression is applied to any signal at or above the threshold.
For example, with a 10 dB knee and a dB threshold, subtle compression will begin at dB and increase so that signals at dB will be fully compressed. The Attack and Release controls are essential parameters for controlling the response time of Compressor by defining how fast it reacts to input-level changes.
Attack defines how long it takes to reach maximum compression once a signal exceeds the threshold, while Release sets how long it takes for the compressor to return to normal operation after the signal falls below the threshold. With Auto Release enabled, the release time will adjust automatically based on the incoming audio. A slight amount of attack time ms allows peaks to come through unprocessed, which helps preserve dynamics by accentuating the initial portion of the signal.
Careful adjustment of attack and release times is essential when it comes to compression of rhythmical sources. If you are not used to working with compressors, play a drum loop and spend some time adjusting Attack, Release, Threshold and Gain.
It can be very exciting! A compressor can only react to an input signal once it occurs. A digital compressor can solve this problem by simply delaying the input signal a little bit. Compressor offers three different Lookahead times: zero ms, one ms and ten ms.
The results may sound pretty different depending on this setting. Compressor can be switched between three basic modes of operation. With Peak selected, Compressor reacts to short peaks within a signal. This mode is more aggressive and precise, and so works well for limiting tasks where you need to ensure that there are absolutely no signals over the set threshold. RMS mode causes Compressor to be less sensitive to very short peaks and compress only when the incoming level has exceeded the threshold for a slightly longer time.
In Expand mode, the Ratio can also be set to values below 1. In this state, Compressor acts as an upward expander, and will increase the gain when signals exceed the threshold. For more information about the various types of dynamics processing, see the Multiband Dynamics chapter see In addition to these modes, Compressor can be switched between two envelope follower shapes that offer further options for how the device measures and responds to signal levels.
In linear Lin mode, the speed of the compression response is determined entirely by the Attack and Release values. In logarithmic Log mode, sharply compressed peaks will have a faster release time than less compressed material.
This can result in smoother and less noticeable compression than Lin mode. Normally, the signal being compressed and the input source that triggers the compressor are the same signal. But by using sidechaining , it is possible to compress a signal based on the level of another signal or a specific frequency component.
To access the Sidechain parameters, unfold the Compressor window by toggling the button in its title bar. The sidechain parameters are divided into two sections.
On the left are the external controls. The sidechain audio is only a trigger for the compressor and is never actually heard. On the right of the external section are the controls for the sidechain EQ. Enabling this section causes the compressor to be triggered by a specific band of frequencies, instead of a complete signal. This section presents some tips for using Compressor effectively, particularly with the sidechain options.
For example, imagine that you have one track containing a voiceover and another track containing background music. Since you want the voiceover to always be the loudest source in the mix, the background music must get out of the way every time the narrator is speaking.
Using the sidechain EQ in conjunction with this technique can create ducking effects even if you only have a mixed drum track to work with as opposed to an isolated kick drum. In this case, insert the Compressor on the track you want to duck.
Then choose the drum track as the external sidechain source. Then enable the sidechain EQ and select the lowpass filter. By carefully adjusting the Frequency and Q settings, you should be able to isolate the kick drum from the rest of the drum mix. Joshua D. Live Sets that use Compressor and which were made in earlier versions of Live will show an Upgrade button in the title bar of each instance of Compressor when loading the Set in Live 9. Press the Upgrade button in order to upgrade that Compressor instance to the latest, improved algorithms.
Note that this may cause your Set to sound different. Corpus is an effect that simulates the acoustic characteristics of seven types of resonant objects. Developed in collaboration with Applied Acoustics Systems, Corpus uses physical modelling technology to provide a wide range of parameters and modulation options. PB Range sets the range in semitones of pitch bend modulation.
With Frequency disabled, the Tune control adjusts the base frequency of the resonance in Hertz. The corresponding MIDI note number and fine tuning offset in cents is displayed below. The slider below the switch determines the extent to which MIDI note off messages mute the resonance.
This is similar to how real-world mallet instruments such as a marimbas and glockenspiels behave. This button will light up if the sidechain is active.
The Amount control sets how much the LFO affects the frequency. It can be set in terms of Hertz, or synced to the song tempo, allowing for controlled rhythmic modulation. Available LFO waveform shapes are sine creates smooth modulations with rounded peaks and valleys , square, triangle, sawtooth up, sawtooth down and two types of noise stepped and smooth. For the noise waveforms, the Phase and Spin controls are not relevant and do not affect the sound.
Spread detunes the two resonators in relation to each other. Positive values raise the pitch of the left resonator while lowering the pitch of the right one, while negative values do the opposite.
The resonance type chooser allows you to select from seven types of physically modeled resonant objects:. The resonator quality chooser controls the tradeoff between the sound quality of the resonators and performance by reducing the number of overtones that are calculated.
This parameter is not used with the Pipe or Tube resonators. The Decay knob adjusts the amount of internal damping in the resonator, and thus the decay time. The Material knob adjusts the variation of the damping at different frequencies. At lower values, low frequency components decay slower than high frequency components which simulates objects made of wood, nylon or rubber.
At higher values, high frequency components decay slower which simulates objects made of glass or metal. The Radius parameter is only available for the Pipe and Tube resonators. Radius adjusts the radius of the pipe or tube. As the radius increases, the decay time and high frequency sustain both increase. At very large sizes, the fundamental pitch of the resonator also changes. The Brightness control adjusts the amplitude of various frequency components. At higher values, higher frequencies are louder.
At negative values, frequencies are compressed, increasing the amount of lower partials. At positive values, frequencies are stretched, increasing the amount of upper partials. Opening, which is only available for the Pipe resonator, scales between an open and closed pipe. The Listening L and R controls adjust the location on the left and right resonator where the vibrations are measured.
Higher values move the listening point closer to the edge. These parameters are not used with the Pipe or Tube resonators, which are always measured in the middle of their permanently open end. The Hit knob adjusts the location on the resonator at which the object is struck or otherwise activated. Higher values move the activation point closer to the edge. The processed signal is fed through a lowpass and highpass filter that can be controlled with an X-Y controller.
To define the filter bandwidth, click and drag on the vertical axis. To set the position of the frequency band, click and drag on the horizontal axis. The filter can be toggled on or off with the Filter switch. Width adjusts the stereo mix between the left and right resonators. Bleed mixes a portion of the unprocessed signal with the resonated signal. At higher values, more of the original signal is applied. This is useful for restoring high frequencies, which can often be damped when the tuning or quality are set to low values.
This parameter is unavailable with the Pipe or Tube resonators. Corpus contains a built-in limiter that automatically activates when the audio level is too high. To refer delay time to the song tempo, activate the Sync switch, which allows using the Delay Time beat division chooser. The numbered switches represent time delay in 16th notes. If the Sync switch is off, the delay time reverts to milliseconds.
In this case, to edit the delay time, click and drag up the Delay Time knob. Internally, they are two independent feedback loops, so a signal on the left channel does not feed back into the right channel and vice versa. The button will cause the delay to endlessly cycle the audio which is in its buffer at the moment that the button is pressed, ignoring any new input until Freeze is disabled. The delay is preceded by a bandpass filter that can be toggled on and off with a switch, and controlled with an X-Y controller.
Filter frequency and delay time can be modulated by an LFO, making it possible to achieve a range of sounds from light chorus-like effects through to heavy contorted noise. The Rate slider sets the frequency of the modulation oscillator in Hertz. The Filter slider adjusts the amount of modulation that is applied to the filter, and the Time slider adjusts the amount of modulation that is applied to the delay time. Changing the delay time while Delay is processing audio can cause abrupt changes in the sound of the delayed signal.
You can choose between three delay transition modes:. Tip: try using the Time slider to explore the effect of time modulation on the different transition modes. Set it to percent when using Delay in a return track. Sets saved in versions of Live older than Live Enable the Stereo Link switch and set the delay time to around ms. Select the Fade transition mode and make sure Ping Pong is disabled. Enable the bandpass filter, set the Filter Frequency to Hz, and adjust the Width slider to 6.
Select the Repitch transition mode and enable the Ping Pong switch. Drum Buss is an analog-style drum processor that was designed to add body and character to a group of drums, while gluing them together in a tight mix. The Trim slider lets you reduce the input level before any processing is applied to the signal. The Comp toggle applies a fixed compressor to the input signal before it is processed by the distortion.
The compressor is optimized for balancing out groups of drums, with fast attack, medium release and moderate ratio settings, as well as ample makeup gain. There are three types of distortion which can be applied to the input signal. Each distortion type adds an increasing degree of distortion, while lending its own character to the overall sound:. For more intensity, it is possible to drive the input prior to distorting it. The Drive control lets you determine how much drive is applied to the input signal.
Drum Buss combines commonly-used drum processing tools for shaping the mid-high range and filling out the low end, which we will look at in the following sections. The mid-high frequency shaping tools are designed to add clarity and presence to drums such as snares and hi-hats. The Damp control is a low-pass filter, which removes unwanted high frequencies that can occur after adding distortion. The Transients knob emphasizes or de-emphasizes the transients of frequencies above Hz.
Negative values also add attack, but decrease sustain. This tightens up the drums, giving them a sharper, more crisp sound with less room and rattle. These tools help you to fill out the low-end of your drums. The Boom knob adjusts the amount of low-end enhancement that the resonant filter produces. The Freq knob adjusts the frequency of the low-end enhancer.
The Decay control adjusts the decay rate of the low frequencies. To solo the result of the low-frequency enhancer, enable Boom Audition via the headphone icon. The Dynamic Tube effect infuses sounds with the peculiarities of tube saturation. An integrated envelope follower generates dynamic tonal variations related to the level of the input signal.
Three tube models, A, B and C, provide a range of distortion characteristics known from real amplifier tubes. Tube A does not produce distortions if Bias is set low, but will kick in whenever the input signal exceeds a certain threshold, creating bright harmonics. Tube C is a very poor tube amp that produces distortions all the time. The qualities of Tube B lie somewhere between these two extremes. The Tone control sets the spectral distribution of the distortions, directing them into the higher registers, or through the midrange and deeper.
The Drive control determines how much signal reaches the tube; greater Drive yields a dirtier output. The intensity of the tube is controlled by the Bias dial, which pushes the signal into the celebrated realms of nonlinear distortion. With very high amounts of Bias, the signal will really start to break apart. The Bias parameter can be positively or negatively modulated by an envelope follower, which is controlled with the Envelope knob.
The more deeply the envelope is applied, the more the Bias point will be influenced by the level of the input signal. Negative Envelope values create expansion effects by reducing distortion on loud signals, while positive values will make loud sounds dirtier. Attack and Release are envelope characteristics that define how quickly the envelope reacts to volume changes in the input signal.
Together, they shape the dynamic nature of the distortions. Note that if Envelope is set to zero, they will have no effect. This improves the sound quality, particularly with high frequency signals, but there is a slight increase in CPU usage.
Echo is a modulation delay effect that lets you set the delay time on two independent delay lines, while giving you control over envelope and filter modulation. The Left and Right delay line controls let you choose the delay time, which can be set in beat divisions or milliseconds, depending on the state of the Sync switch.
You can use the Sync Mode choosers to select one of the following beat-synced modes: Notes, Triplet, Dotted and 16th. Note that when switching between Sync Modes, the resulting changes are only audible while the Sync switch is set to Sync. Note that when Stereo Link is enabled, the Delay Offset can still be adjusted individually for the two delay lines. The Input knob sets the amount of gain applied to the dry signal. You can adjust the delay times for each delay line by clicking and dragging in the display.
The Filter toggle enables a high-pass and low-pass filter. The Filter Display makes it possible to visualize the filter curves. To show or hide the Filter Display, use the triangular toggle button. You can also adjust the filter parameters by clicking and dragging either of the filter dots in the Filter Display. You can choose from one of six different modulation waveforms including sine, triangle, sawtooth up, sawtooth down, square, and noise. The selected waveform will appear in the display, which you can drag to adjust the modulation frequency.
When Sync is enabled, modulation is synchronized to the song tempo. You can use the Rate slider to set the frequency of the modulation oscillator in beat divisions. When Sync is disabled, you can use the Freq slider to adjust frequency of the modulation oscillator in Hertz. Phase adjusts the amount of offset between the waveforms for the left and right channel. Mod Delay adjusts the amount of modulation that is applied to the delay time.
Modulation x4 scales the delay time modulation depth by a factor of four. With short delay times, this produces deep flanging sounds. Env Mix blends between the modulation oscillator and an envelope follower. It mutes the signal components below its threshold. Threshold sets the threshold level that incoming audio signals must exceed in order to open the gate. Release sets how long it takes for the gate to close after the signal has dropped below the threshold. When Ducking is enabled, the wet signal is proportionally reduced as long as there is an input signal.
Ducking begins to affect the output signal when the input level exceeds the set Threshold. Release sets how long it takes for ducking to stop after the input signal drops below the threshold.
When enabled, Noise introduces noise to simulate the character of vintage equipment. You can adjust the Amount of noise added to the signal, and Morph between different types of noise. When enabled, Wobble adds an irregular modulation of the delay time to simulate tape delays. You can adjust the Amount of wobble added to the signal, and Morph between different types of wobble modulation. Repitch causes a pitch variation when changing the delay time, similar to the behavior of hardware delay units.
When Repitch is disabled, changing the delay time creates a crossfade between the old and new delay times. Note that in order to save CPU, the Echo device turns itself off at least eight seconds after its input stops producing sound.
However, Echo will not switch off if both the Noise and Gate parameters are enabled. The Reverb knob sets the amount of reverb added, and you use the Reverb Location chooser to set where the reverb is added in the processing chain: pre delay, post delay, or within the feedback loop.
Use the Decay slider to lengthen or shorten the reverb tail. The Stereo control sets the stereo width of the wet signal. The Output sets the amount of gain applied to the processed signal. Set it to percent when using Echo in a return track. Stereo mode uses a single curve to filter both channels of a stereo input equally. In all modes, the frequency spectrum of the output is displayed behind the filter curves when the Analyze switch is on.
The Edit switch indicates the active channel, and is used to toggle between the two curves. Each filter has a chooser that allows you to switch between eight responses. From top to bottom in the choosers, these are:. Each filter band can be turned on or off independently with an activator switch below the chooser. Turn off bands that are not in use to save CPU power.
To achieve really drastic filtering effects, assign the same parameters to two or more filters. To edit the filter curve, click and drag on the filter dots in the display.
Note that the gain cannot be adjusted for the low cut, notch and high cut filters. In these modes, vertical dragging adjusts the filter Q. When using this expanded view, all eight filters can be edited simultaneously in the Device View.
Sound clips are grouped together on Sound Cloud Warning: unmastered, so the volume fluctuates from clip to clip. Individual descriptions below. He offers an amazing range of free virtual effects and synthesizers. They are all worth checking out, but the TAL-Dub vintage delay effect is a must-have. Using the Damp and Resonance knobs, you can evoke classic tape and dub effects, but with two independent or linked delay lines, it offers much more than retro sounds.
Sound Clip: I just played one distorted chord. Ignite makes hardware amplifiers and effects, as well as virtual amplifier and effect plug-ins. This overdrive plug-in is natural sounding and dynamic. First you will hear it without the and then with. Note how the distortion reacts to the strength of my picking. I can play clean chords by just lightening up. Michael Norris is a Wellington-based composer, software developer and music theorist.
His free suite of plug-ins offers a wide range of effects currently unavailable with hardware. For example: Spectral delays allow you to delay a particular frequency range of a signal while leaving the rest of the spectrum dry. They allow enormous audio processing control and creativity, from audio files or live audio.
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