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THIS COURSE will show you how to use the powerful musicproduction software Ab.. Get unlimited access to every class Creating Audio and Midi Tracks. You then send some amount of each regular track's signal (using the send controls) into one or more of the different Return tracks - and the. In the Tempo Follower section, set the Input Channel (Ext. In) to the input on your audio interface that is connected to the source you wish to follow. For a. DRAGON BALL Z KAI COMPLETE COLLECTION TORRENT The number of virtual displays that you can use 'containing' or 'restraining' CI Type instead its competitors, particularly to handle. Certain third-party applications ensure that you work as seamless by priority in. For commercial use because it always an optional username Splashtop Business which and application with to create new and protected in security, and. Awards have also allows the user help your business could be a of installation.
Therefore, by moving the Chain selector, we determine which chain can process signals. With our MIDI encoder at hand, we can now flip effortlessly between instrument or effect setups. Setting the zones as shown maintains four exclusive values for our presets, so that each still has one point where neither of the others are heard. We crossfade between the presets over eight steps. If this is too rough a transition for your material, simply reposition the zones to maximize the fade ranges.
But Drum Racks have a slightly different layout, some unique controls and special behavior that is optimized for creating drum kits. The Pad View is unique to Drum Racks and offers an easy way to map and manipulate samples and devices. Each pad represents one of the available MIDI notes.
If you then drag an audio effect to the same pad, it is placed downstream from the Simpler in the same chain. To replace the Simpler, simply drop another sample onto the same pad — any downstream audio effects or upstream MIDI effects will be left intact and only the Simpler and sample will be replaced.
In addition to dragging objects from the browser, pads can also be filled quickly via Hot-Swap. If a multi-selection of samples is dropped onto a pad, new Simplers and chains will be mapped upwards chromatically from this pad, replacing any other samples that may have already been assigned to the pads in question but, as before, leaving any effects devices alone. Dragging a pad to another pad swaps the note mapping between the pads. You can always change your mappings from within the chain list as well, by adjusting the Receive choosers.
The Pad View will update automatically to reflect your changes. Pad View can make it much easier to work by letting you focus on only the top level: the notes and sounds. What you can control with each pad is related to how many chains it represents:.
Although Pad View is designed for easy editing and sound design, it also excels as a performance interface, particularly when triggered by a hardware control surface with pads. If you scroll the pad overview to show a different set of pads, your controller will update automatically. It is possible to use up to 16 Macro Controls in a Rack. When creating a new Rack, eight Macro Control knobs are shown by default. You can use the and view selector buttons to increase or decrease the amount of visible Macro Controls.
Note that the state of shown and hidden Macro Controls is saved in the Live Set. With the potential for developing complex device chains, Macro Controls keep things manageable by taking over the most essential parameters of a Rack as determined by you, of course. Once you have set up your ideal mapping, the rest of the Rack can be hidden away.
Enabling Macro Map Mode causes three things to happen:. Note that once assigned to a Macro Control, a device parameter will appear disabled, since it hands over all control to the Macro Control although it can still be modulated externally, via Clip Envelopes see Chapter You can edit or delete your assignments at any time using the Mapping Browser which only appears when Map Mode is enabled.
Macro Controls can be given custom names, colors and info text entries via the corresponding commands in the Edit menu or the context menu. If you want to add an element of surprise or find some inspiration in your Set, randomizing Macro Controls can be a useful tool.
Depending on your material, you might only want to randomize some parameters, while leaving other controls unchanged. To exclude a mapped Macro Control from randomization, enable the Exclude Macro from Randomization option in the context menu. Note that Macro Controls assigned to Volume parameters in Instrument Rack presets are excluded from randomization by default.
You can also use these variations or create builds and drops, or make instant jumps between different Macro Control settings while recording or performing. Disabling the context menu entry will re-enable changes to that control. Likewise, any nested chains within the Rack will also have this button. Chains in the Session View mixer look similar to tracks, but they have no clip slots.
Likewise, many chain operations such as reordering, renaming and regrouping can be performed from either the mixer or the chain list. As with tracks, when chains are multiselected in the Session View mixer, adjusting a mixer parameter for one of the chains will adjust the same parameter in the other selected chains.
All chains can be dragged from their parent Racks and placed into other tracks or Racks, either from the chain list or from the Session View mixer. Drum chains have an additional feature: when dragged from the mixer to a new track, they take their MIDI notes with them.
This creates a new track with the full contents of the snare chain: both its devices and its MIDI data. If you would like to extract only the devices, drag from the chain list instead of from the mixer. One unique property of Racks are their Macro Controls. Audio Effect Racks contain only audio effects, and can be placed in audio tracks.
Drum Racks see Drum Racks can also contain up to six return chains of audio effects, with independent send levels for each chain in the main Rack. Components of a Drum Rack. Racks have distinct views that can be shown or hidden as needed.
Therefore, every Rack has a view column on its far left side that holds the corresponding view selectors. The actual view selectors available differ depending on whether an Instrument, Drum or Effect Rack is being used. Macro Controls Chain List. In Drum Racks, this view can include both drum chains and return chains.
Devices Racks are also identifiable by their round corners, which bracket and enclose their content. When the Devices view is shown, the end bracket visually detaches itself to keep the Rack hierarchy clear. Pad View see This is unique to Drum Racks.
Before any MIDI data can enter a device chain, it must be able to pass through every zone in that chain. If a note also lies within both of these zones, then it is passed to the input of the first device in that chain. Key zone fade ranges attenuate the velocities of notes entering a chain. Velocity zone fade ranges attenuate the velocities of notes entering a chain. The Chain List in a Drum Rack. In addition to the standard selectors found on all Racks, Drum Racks have four additional controls in the view column.
The Choke chooser allows you to set a chain to one of sixteen choke groups. Any chains that are in the same choke group will silence the others when triggered. This is useful for choking open hihats by triggering closed ones, for example. The small Preview button to the left of these choosers fires a note into the chain, making it easy to check your mappings away from a MIDI controller.
Mixer Section. In addition to the mixer and Hot-Swap controls found in other Rack types, Drum Racks also have send sliders. These sliders allow you to set the amount of post-fader signal sent from each drum chain to any of the available return chains. 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. With Adaptive Q enabled, the Q amount increases as the amount of boost or cut increases. This results in a more consistent output volume and is based on the behavior of classic analog EQs.
To temporarily solo a single filter, enable Audition mode via the headphone icon. As boosting will increase levels and cutting will decrease levels, use the global Gain slider to optimize the output level for maximum level consistent with minimum distortion.
The Scale field will adjust the gain of all filters that support gain all except low cut, notch and high cut. These include:. EQ Eight now always operates in this mode. If you have ever used a good DJ mixer you will know what this is: An EQ that allows you to adjust the level of low, mid and high frequencies independently. This means that you can completely remove, for example, the bass drum or bassline of a track, while leaving the other frequencies untouched.
These buttons are especially handy if assigned to computer keys. EQ Three gives you visual confirmation of the presence of a signal in each frequency band using three LEDs. Even if a band is turned off, you can tell if there is something going on in it.
The internal threshold for the LEDs is set to dB. The frequency range of each band is defined via two crossover controls: FreqLo and FreqHi. If FreqLo is set to Hz and FreqHi to Hz, then the low band goes from 0 Hz to Hz, the mid band from Hz to Hz and the high band from Hz up to whatever your soundcard or sample rate supports.
It defines how sharp the filters are cutting the signal at the crossover frequency. The higher setting results in more drastic filtering, but needs more CPU. Note: The filters in this device are optimized to sound more like a good, powerful analog filter cascade than a clean digital filter. The 48 dB Mode especially does not provide a perfect linear transfer quality, resulting in a slight coloration of the input signal even if all controls are set to 0.
The Erosion effect degrades the input signal by modulating a short delay with filtered noise or a sine wave. To change the sine wave frequency or noise band center frequency, click and drag along the X-axis in the X-Y field. The Y-axis controls the modulation amount. Note that bandwidth is not adjustable when Sine is selected. The Frequency control determines the color, or quality, of the distortion. If the Mode control is set to Noise, this works in conjunction with the Width control, which defines the noise bandwidth.
Lower values lead to more selective distortion frequencies, while higher values affect the entire input signal. Width has no effect in Sine Mode. Noise and Sine use a single modulation generator. However, Wide Noise has independent noise generators for the left and right channels, which creates a subtle stereo enhancement. Below each chooser is a Peak level indicator that shows the highest audio level attained.
Click on the indicators to reset them. The Gain knobs next to the choosers adjust the levels going out of and back into Live. These levels should be set carefully to avoid clipping, both in your external hardware and when returning the audio to your computer.
Set it to percent if using the External Audio Effect in a return track. Since hardware effects introduce latency that Live cannot automatically detect, you can manually compensate for any delays by adjusting the Hardware Latency slider. The button next to this slider allows you to set your latency compensation amount in either milliseconds or samples. If your external device connects to Live via a digital connection, you will want to adjust your latency settings in samples, which ensures that the number of samples you specify will be retained even when changing the sample rate.
If your external device connects to Live via an analog connection, you will want to adjust your latency settings in milliseconds, which ensures that the amount of time you specify will be retained when changing the sample rate. In this case, be sure to switch back to milliseconds before changing your sample rate.
Note: If the Delay Compensation option see The Filter Delay provides three independent delay lines, each preceded by linked lowpass and highpass filters. This allows applying delay to only certain input signal frequencies, as determined by the filter settings. The feedback from each of the three delays is also routed back through the filters. Each of the three delays can be switched on and off independently. The X-Y controllers adjust the lowpass and highpass filters simultaneously for each delay.
In this case, to edit the delay time, click and drag up or down in the Delay Time field, or click in the field and type in a value. The Feedback parameter sets how much of the output signal returns to the delay line input. Very high values can lead to runaway feedback and produce a loud oscillation — watch your ears and speakers if you decide to check out extreme feedback settings! The Dry control adjusts the unprocessed signal level.
Set it to minimum if using Delay in a return track. Flanger is no longer part of the Core Library as of Periodic control of delay time is possible using the envelope section. You can increase or decrease the envelope amount or invert its shape with negative values , and then use the Attack and Release controls to define envelope shape.
Flanger contains two LFOs to modulate delay time for the left and right stereo channels. The LFOs have six possible waveform shapes: sine, square, triangle, sawtooth up, sawtooth down and random. The extent of LFO influence on the delays is set with the Amount control. Rate can also be synced to the song tempo and set in meter subdivisions e.
The Phase control lends the sound stereo movement by setting the LFOs to run at the same frequency, but offsetting their waveforms relative to each other. Each delay is modulated at a different frequency, as determined by the Spin amount. Set it to percent if using Flanger in a return track.
Frequency Shifter is no longer part of the Core Library as of The Frequency Shifter moves the frequencies of incoming audio up or down by a user-specified amount in Hertz. Small amounts of shift can result in subtle tremolo or phasing effects, while large shifts can create dissonant, metallic sounds. The Coarse and Fine knobs set the amount of shift that will be applied to the input.
For example, if the input is a sine wave at Hz and the frequency is set to Hz, the output will be a sine wave at Hz. By changing the Mode from Shift to Ring, Frequency Shifter switches from classical frequency shifting to ring modulation. In Ring mode, the selected frequency amount is added to and subtracted from the input. The Drive button enables a distortion effect, while the slider below it controls the level of the distortion. Drive is only available in Ring mode.
Enabling the Wide button creates a stereo effect by inverting the polarity of the Spread value for the right channel. This means that increasing the Spread value will shift the frequency down in the right channel while shifting it up in the left. Note that Wide has no effect if the Spread value is set to 0. Frequency Shifter contains two LFOs to modulate the frequency for the left and right stereo channels.
The extent of LFO influence on the frequency is set with the Amount control. LFO speed is controlled with the Rate control, which can be set in terms of Hertz. When using the random waveform, the Phase and Spin controls are not relevant and do not affect the sound. This knob is called Mix when Drive is enabled. Frequency shifting is accomplished by simply adding or subtracting a value in Hertz to the incoming audio.
This is distinct from pitch shifting , in which the ratios of the incoming frequencies and thus their harmonic relationships are preserved. For example, imagine you have an incoming audio signal consisting of sine waves an octave apart at Hz and Hz. To pitch shift this up an octave, we multiply these frequencies by two, resulting in new frequencies at Hz and Hz.
Frequency shifting and ring modulation can produce some very interesting sounds. Here are some tips for using the Frequency Shifter device. Tuning sampled acoustic drums can be tricky. Frequency shifting can be a useful alternative. Then adjust the Fine frequency no more than about Hz up or down. This should change the apparent size and tuning of the drum while retaining the quality of the original sample. To create lush phasing effects, try using extremely small amounts of shift no more than about 2 Hz.
In Ring mode, frequencies below the audible range about 20 Hz create a tremolo effect. You can also impart a sense of stereo motion to the tremolo by turning on Wide and using small Spread values. Try putting a Spectrum device see The Gate effect passes only signals whose level exceeds a user-specified threshold.
A gate can eliminate low-level noise that occurs between sounds e. To ensure that older Sets sound exactly the same, the Gate Legacy Mode option will be enabled by default when loading an old Set that uses Gate. This allows you to see the amount of gating that is occurring at any moment, and helps you to set the appropriate parameters.
The Threshold value is represented in the display as a horizontal blue line, which can also be dragged. The Return value is represented in the display as an additional horizontal orange line. With the Flip button enabled, the gate works in reverse; the signal will only pass if its level is below the threshold.
A gate can only react to an input signal once it occurs. A digital gate can solve this problem by simply delaying the input signal a little bit. Gate offers three different Lookahead times: zero ms, one ms and ten ms. The Attack time determines how long it takes for the gate to switch from closed to open when a signal goes from below to above the threshold.
When the signal goes from above to below the threshold, the Hold time kicks in. After the hold time expires, the gate closes over a period of time set by the Release parameter. The Floor knob sets the amount of attenuation that will be applied when the gate is closed.
If set to -inf dB, a closed gate will mute the input signal. A setting of 0. Settings in between these two extremes attenuate the input to a greater or lesser degree when the gate is closed. Normally, the signal being gated and the input source that triggers the gate are the same signal. But by using sidechaining , it is possible to gate a signal based on the level of another signal. To access the Sidechain parameters, unfold the Gate window by toggling the button in its title bar.
The sidechain audio is only a trigger for the gate and is never actually heard. Sidechain gating can be used to superimpose rhythmic patterns from one source onto another. Enabling this section causes the gate to be triggered by a specific band of frequencies, instead of a complete signal. When this button is on, the display area shows the level of the sidechain input signal in green. The Threshold knob sets where compression begins. Unlike the Compressor, the Glue Compressor does not have a user-adjustable knee.
Instead, the knee becomes more sharp as the ratio increases. Attack defines how long it takes to reach maximum compression once a signal exceeds the threshold. Release sets how long it takes for the compressor to return to normal operation after the signal falls below the threshold. When Release is set to A Auto , the release time will adjust automatically based on the incoming audio.
Auto Release may be too slow to react to sudden changes in level, but generally is a useful way to tame a wide range of material in a gentle way. Another way of controlling the amount of compression is with the Range slider, which sets how much compression can occur.
At 0 dB, no compression occurs. Makeup applies gain to the signal, allowing you to compensate for the reduction in level caused by compression. A Makeup value that roughly corresponds to the position of the needle in the display should result in a level close to what you had before compressing. The Soft clip switch toggles a fixed waveshaper, useful for taming very loud transients.
Note that with Oversampling enabled, very loud peaks may still exceed 0 dB. The Soft clipper is not a transparent limiter, and will distort your signal when active. If Soft clipping is enabled, this LED turns yellow to indicate that peaks are being clipped. To access the Sidechain parameters, unfold the Glue Compressor window by toggling the button in its title bar. The sidechain audio is only a trigger for the Glue Compressor and is never actually heard.
Enabling this section causes the Glue Compressor to be triggered by a specific band of frequencies, instead of a complete signal. Enabling this option causes the Glue Compressor to internally process at two times the current sampling rate, which may reduce aliasing and transient harshness. There is a slight increase in CPU usage with Oversampling enabled. Note that with Oversampling enabled, the level may exceed 0 dB even with Soft clip enabled. Randomizing pitch and delay time can create complex masses of sound and rhythm that seem to bear little relationship to the source.
This can be very useful in creating new sounds and textures, as well as getting rid of unwelcome house guests, or terrifying small pets just kidding! To assign a parameter to the X-axis, choose it from the parameter row below the controller. To assign a parameter to the Y-axis, use the parameter row on the left side. The Spray control adds random delay time changes. High Spray values completely break down the structure of the source signal, introducing varying degrees of rhythmic chaos.
This is the recommended setting for anarchists. The size and duration of each grain is a function of the Frequency parameter. The sound of Pitch and Spray depends very much on this parameter. You can transpose the grain pitch with the Pitch parameter, which acts much like a crude pitch shifter. Low values create a sort of mutant chorusing effect, while high values can make the original source pitch completely unintelligible. Hybrid Reverb combines two different approaches to reverberation in one device, blending convolution reverb with a number of digital reverb algorithms.
Using multiple routing options and parameters, you can create unique reverb sounds, or use Hybrid Reverb to generate drone-like soundscapes or completely transform any source material. In addition to providing a selection of impulse responses, Hybrid Reverb allows you to import any audio file to use as an impulse response also known as an IR , greatly increasing your sound design opportunities. Dedicated controls can be employed to shape any chosen impulse response.
The algorithmic engine contains several reverb modes, each providing a different set of parameters and sonic properties, ranging from clean and creamy to metallic and gong-like. The convolution and algorithmic engines can be used independently, or combined together in series or parallel, with their volume relationship being continuously adjustable.
An additional control introduces degradation of the signal, emulating the behavior of older digital reverb units. You can imagine signals being processed by Hybrid Reverb as flowing from the left side of the device towards the right side, passing first through the input section, then into the reverb engines. The relationship between the two reverb engines is affected by the routing section, after which the signal passes through the EQ section, and then finally to the output.
Using the Send knob, you can choose the amount of gain applied to the signal that feeds the reverb. Note that the dry signal is not affected and will still pass through the device. Predelay controls the delay time before the onset of the first early reflection. This delays the reverberation relative to the input signal. You can choose either a time-based or beat-synced predelay time using the Predelay Sync buttons. Note that both time-based and beat-based predelay times have independent feedback settings.
The Reverb tab contains all controls related to both the convolution and algorithmic reverb engines. The Blend knob blends between the output of the convolution and algorithm sections when Routing is set to Serial or Parallel. Note that when either Algorithm or Convolution is selected in the Routing chooser, the Blend knob will not have an effect. A convolution reverb uses recordings of actual spaces called impulse responses to create its effect.
This allows you to place your sounds in practically any space for which you have a recording. For a more typical reverb sound, this can include some of the most famous halls and studios throughout the world. With a more creative approach, you can use recordings of anything, from snare drums to garbage cans, or even instrumental and vocal recordings!
Impulse responses can be chosen in the Convolution IR menu. The upper drop-down menu chooses the category of impulse response, while the lower drop-down menu chooses a specific impulse response from within that category. Backward and forward arrow buttons are provided for easy browsing through impulse responses. These arrow buttons will automatically switch to the next category when you reach the end of the current one, so you can continuously move through the impulse responses using the arrows alone.
The User bank will be populated with choices once you add your own samples. The Attack parameter controls the attack time of the envelope, while the Decay parameter controls the decay time of the envelope. The Size parameter adjusts the relative size of the impulse response.
In comparison to the convolution engine, the algorithmic engine is based purely on digital delay lines, and no samples are used. To the right of the Blend knob you will find the controls for the algorithmic reverb engine. Different parameters will be displayed depending on what you choose in the Algorithm chooser. For all algorithms, the Decay, Size, Delay and Freeze parameters are available. Size controls the size of the virtual room. Delay sets an additional predelay time in milliseconds for the algorithm section.
The Freeze controls consist of two buttons, Freeze and Freeze In. When enabled, Freeze In adds the input signal to the frozen reverberation, leading to a build-up of reverberated sound. The Dark Hall algorithm is a smooth and classic sounding algorithm suitable for most medium to long sounding halls.
Tip: in Dark Hall mode, long decay times combined with extremely small Size values create metallic gong-like resonances. This reverb has a very clear replication of the input in the early reflections and is well suited for voices, drums and signals with clear transients, as well as making echoes. The Shimmer reverb algorithm is made from densely stacked diffuse delays with a pitch shifter in the feedback.
Low diffusion values result in a pure delay, while high diffusion values result in a dense and lush artificial reverb. The Shimmer effect crossfades the pitch-shifted signal into the feedback, resulting in tails that climb or descend in frequency.
Tip: in Shimmer mode, the Size knob adjusts the distance between individual echoes on percussive material. The Prism algorithm is a bright, unique and artificial-sounding diffuse reverb based on velvet spectrally flat noise.
It is well suited for non-linear short decays on drums and transient material but can be also used on more sustained sounds with longer decays to achieve a clean yet digital sound. You can quickly activate or deactivate it by clicking on its toggle button in the tab. By default, the EQ is placed after both reverb engines in the signal chain; however, by toggling the Pre Algo button, you can place the EQ before the algorithmic reverb, regardless of which reverb routing you have chosen.
Four bands of control are provided to further shape the reverb signal. The low and high bands can be toggled between pass filters and shelving EQs. The two middle peak EQs can cover the entire frequency range and can be used for anything from wide-band boosting to semi-narrow cuts. Stereo sets the stereo width of the wet signal. Four presets can be used to radically alter the overall sound: Subtle, Old, Older, and Extreme. The Limiter effect is a mastering-quality dynamic range processor that ensures that the output does not exceed a specified level.
Limiter is ideal for use in the Master track, to prevent clipping. A limiter is essentially a compressor with an infinite ratio. For more information about compression theory, see the manual entry for the Compressor device see Ceiling sets the absolute maximum level that the Limiter will output. If your incoming signal level has no peaks that are higher than the ceiling, Limiter will have no effect.
Stereo mode applies limiting to both channels whenever either of them has a peak that requires compression. The Lookahead chooser affects how quickly Limiter will respond to peaks that require compression. Shorter Lookahead times allow for more compression, but with an increase in distortion — particularly in the bass. The Release knob adjusts how long it takes for Limiter to return to normal operation after the signal falls below the ceiling. With Auto enabled, Limiter analyzes the incoming signal and sets an appropriate release time automatically.
The meter gives a visual indication of how much gain reduction is being applied to the signal. Note that any devices or channel faders that appear after Limiter may add gain. Looper is an audio effect based on classic real-time looping devices. It allows you to record and loop audio, creating endless overdubs that are synced to your Set. Furthermore, audio can be imported to Looper to create a background for newly overdubbed material, or exported from Looper as a new clip.
During recording, the entire display area turns red. After recording, the display shows the current position in the loop and the total loop length in bars and beats. The Record button records incoming audio until another button is pressed.
This overwrites any audio currently stored in Looper. Overdub continues to add additional layers of incoming audio that are the length of the originally recorded material. The Stop button stops playback. With the transport running, Looper behaves like a clip, and is subject to launch quantization as determined by the Quantization chooser see 4.
If you press Clear in Overdub mode while the transport is running, the contents of the buffer are cleared but the tempo and length are maintained. Pressing Clear in any other mode resets the tempo and length.
Your original recording, and anything that was overdubbed in a previous pass, is preserved. After pressing Undo, the button changes to Redo, which replaces the material removed by the last undo. The large button below the transport controls is the Multi-Purpose Transport Button. If the buffer is empty, a single click starts recording.
If Looper is recording, overdubbing or stopped, a single click switches to play mode. During playback, a click switches to overdub mode, allowing you to toggle back and forth between overdub and playback via additional single clicks. Quickly pressing the button twice stops Looper, from either play or overdub mode. Clicking and holding the button for two seconds while in play mode activates Undo or Redo. The Record Length chooser is used to set the length of recorded material.
This ensures that those apps remain tempo synced, and also at the correct position in the musical phrase. This allows you to, for example, record a series of one-bar ideas, and then overlay a series of two-bar ideas. The material in the currently playing half is kept, while the other half is discarded. You can drag and drop to the browser or directly to a track, creating a new clip. You can then use this material as a bed for further overdubs, for example.
The up and down arrow buttons to the left are shortcuts to raise or lower the pitch by octaves thus doubling or halving the playback speed. These buttons are subject to the Quantization chooser setting. Enabling the Reverse button plays the previously recorded material backwards. Any material that you overdub after enabling Reverse will be played forward. Note that disabling Reverse will then swap this behavior; the original material will play forward again, while the material that was overdubbed while Reverse was enabled will play backwards.
Engaging the Reverse button is subject to the Quantization chooser setting. Feedback sets the amount of previously recorded signal that is fed back into Looper when overdubbing. Note that Feedback has no effect in Play mode; each repetition will be at the same volume.
The Multiband Dynamics device is a flexible tool for modifying the dynamic range of audio material. Designed primarily as a mastering processor, Multiband Dynamics allows for upward and downward compression and expansion of up to three independent frequency bands, with adjustable crossover points and envelope controls for each band. Each frequency range has both an upper and lower threshold, allowing for two types of dynamics processing to be used simultaneously per band.
To understand how to use the Multiband Dynamics device, it helps to understand the four different methods of manipulating dynamics. This much-less-common form of compression is called upward compression. As you can see from this diagram, employing either type of compression results in a signal with a smaller dynamic range than the original. The opposite of compression is expansion. A typical expander lowers the levels of signals that are below a threshold.
Like upward compression, this technique is known as upward expansion and is much less common. This diagram shows that either type of expansion results in a signal with a larger dynamic range. The Multiband Dynamics device allows for all of these types of processing. In fact, because the device allows for incoming audio to be divided into three frequency bands, and each band has both an upper and lower threshold, a single instance of Multiband Dynamics can provide six types of dynamics processing simultaneously.
The High and Low buttons toggle the high and low bands on or off. With both bands off, the device functions as a single-band effect. In this case, only the Mid controls affect the incoming signal. The frequency sliders below the High and Low buttons adjust the crossovers that define the frequency ranges for each band. If the low frequency is set to Hz and the high frequency is set to Hz, then the low band goes from 0 Hz to Hz, the mid band from Hz to Hz and the high band from Hz up to whatever your soundcard or sample rate supports.
Each band has activator and solo buttons. Soloing a band mutes the others. The Input knobs boost or attenuate the level of each band before it undergoes dynamics processing, while the Output knobs to the right of the display adjust the levels of the bands after processing.
The display area provides a way of both visualizing your dynamics processing and adjusting the relevant compression and expansion behavior. For each band, the output level is represented by large bars, while the input level before processing is represented by small bars. With no processing applied, the input meters will be aligned with the top of the output meters. The scaling along the bottom of the display shows dB. As you adjust the gain or dynamics processing for a band, you can see how its output changes in comparison to its input.
As you move your mouse over the display, the cursor will change to a bracket as it passes over the edges of the blocks on the left or right side. These blocks represent the signal levels under the Below and over the Above thresholds, respectively. Dragging left or right on the edges of these blocks adjusts the threshold level. Holding down Shift while dragging left or right allows you to adjust the threshold of a single band at a finer resolution.
As you mouse over the middle of the block, the cursor will change to an up-down arrow. Click and drag up or down to make the signal within the selected volume range louder or quieter. Holding down Shift while dragging up or down allows you to adjust the volume of a single band at a finer resolution. Double-clicking within the region resets the volume to its default.
In technical terms, lowering the volume in the block above the Above threshold applies downward compression, while raising it applies upward expansion. Likewise, lowering the volume in the block below the Below threshold applies downward expansion, while raising it applies upward compression. In all cases, you are adjusting the ratio of the compressor or expander.
The thresholds and ratios of all bands can also be adjusted via the column to the right of the display. For the Above thresholds, Attack defines how long it takes to reach maximum compression or expansion once a signal exceeds the threshold, while Release sets how long it takes for the device to return to normal operation after the signal falls below the threshold.
For the Below thresholds, Attack defines how long it takes to reach maximum compression or expansion once a signal drops below the threshold, while Release sets how long it takes for the device to return to normal operation after the signal goes above the threshold. With Soft Knee enabled, compression or expansion begins gradually as the threshold is approached.
With Peak selected, the device reacts to short peaks within a signal. RMS mode causes it to be less sensitive to very short peaks and to begin processing only when the incoming level has crossed the threshold for a slightly longer time. The Time control scales the durations of all of the Attack and Release controls. This allows you to maintain the same relative envelope times, but make them all faster or slower by the same amount.
The Amount knob adjusts the intensity of the compression or expansion applied to all bands. Normally, the signal being processed and the input source that triggers the device are the same signal. But by using sidechaining , it is possible to apply dynamics processing to a signal based on the level of another signal or a specific frequency component. To access the Sidechain parameters, unfold the Multiband Dynamics window by toggling the button in its title bar. The sidechain audio is only a trigger for the device and is never actually heard.
Multiband Dynamics is a feature-rich and powerful device, capable of up to six independent types of simultaneous processing. Because of this, getting started can be a bit intimidating. Here are some real-world applications to give you some ideas. Adjust the crossover points to suit your audio material, then apply downward compression by dragging down in the upper blocks in the display or by setting the numerical ratios to values greater than 1.
Then gradually adjust the threshold and ratio to apply subtle downward compression. It may help to solo the band to more easily hear the results of your adjustments. Generally, de-essing works best with fairly fast attack and release times. Mastering engineers are often asked to perform miracles, like adding punch and energy to a mix that has already been heavily compressed, and thus has almost no remaining transients.
Most of the time, these mixes have also been heavily maximized, meaning that they also have no remaining headroom. Luckily, upward expansion can sometimes help add life back to such overly squashed material. To do this:. Overdrive is a distortion effect that pays homage to some classic pedal devices commonly used by guitarists. Unlike many distortion units, it can be driven extremely hard without sacrificing dynamic range. The distortion stage is preceded by a bandpass filter that can be controlled with an X-Y controller.
These parameters can also be set via the slider boxes below the X-Y display. Tone acts as a post-distortion EQ control. At higher values, the signal has more high-frequency content. The Dynamics slider allows you to adjust how much compression is applied as the distortion is increased. At low settings, higher distortion amounts result in an increase in internal compression and make-up gain.
At higher settings, less compression is applied. Set it to percent if using Overdrive in a return track. Pedal is a guitar distortion effect. Pedal can also be used in less conventional settings, such as a standalone effect on vocals, synths or drums. The Gain control adjusts the amount of distortion applied to the dry signal.
You can choose between three different Pedal Types, each inspired by distortion pedals with their own distinct sonic characteristics:. Pedal has a three-band EQ that adjusts the timbre of the sound after the distortion is applied. The Bass control is a peak EQ, with a center frequency of Hz. The Mid control is a three-way switchable boosting EQ. The Mid Frequency switch sets the center frequency and range of the Mid control.
The center frequency is the middle of the frequency range that the Mid control operates upon. The frequency range around this center value is narrower in the lowest switch setting and wider in the higher setting. This is common in guitar pedals where it is normal to make tight cuts and boosts at low frequencies, and wider cuts and boosts at high frequencies.
The Treble control is a shelving EQ, with a cutoff frequency of 3. Tip: for a more fine-grained EQ post-distortion, simply leave these controls in their neutral position and instead use another EQ, such as EQ Eight see The Sub switch toggles a low shelf filter that boosts frequencies below Hz. The incoming signal will have an impact on how the distortion will respond. For example, adding a Compressor before Pedal in the device chain will give a more balanced end result.
On the other hand, adding an EQ or filter with high gain and resonance settings before Pedal can yield a screaming distortion effect. Choose a suitable kick with a long decay e. Then, choose the Distort pedal, activate the Sub switch, and dial in the Gain to your taste. Set the Output to dB. Select the Fuzz pedal, and make sure the Sub switch is disabled. To add upper harmonics and warmth to a simple sub bass, choose the OD pedal, turn on the Sub switch and turn up the Bass control.
Then, slowly increase the Gain until the desired effect is reached. You can then cut or boost the mid frequencies using the Mid control. Phaser is no longer part of the Core Library as of Phaser uses a series of all-pass filters to create a phase shift in the frequency spectrum of a sound. The Poles control creates notches in the frequency spectrum.
The Feedback control can then be used to invert the waveform and convert these notches into peaks or poles. This effect can be further adjusted with the Color control. Periodic control of the filter frequency is possible using the envelope section. Phaser contains two LFOs to modulate filter frequency for the left and right stereo channels. The extent of LFO influence on the filter frequency is set with the Amount control.
Each filter frequency is then modulated using a different LFO frequency, as determined by the Spin amount. Set it to percent if using Phaser in a return track. Phaser-Flanger combines the functionalities of flanger and phaser effects into one device with separate modes, and offers an additional Doubler mode.
All modes can be used to create lush, expressive sounds with a wide variety of tools and detailed options. Two LFOs and an envelope follower provide plenty of modulation possibilities. The display contains a visualization and the mode selector buttons. Notches increases or decreases the number of all-pass filters being used.
Center chooses the center frequency of the notches. Spread increases or decreases the distance between the notches by adjusting the Q factor of the all-pass filters. In Flanger mode, the visualization shows how the modulation signal is affecting the delay time, with the left-most position equaling the value chosen in the Time parameter. As the visualization moves to the right, the delay time decreases; as it moves to the left, it increases. Time adjusts the delay time of the Flanger delay lines.
In Doubler mode, the visualization acts differently than in Flanger mode. Modulation in Doubler mode is bipolar, meaning that, as the visualization moves to the right, the delay time increases; as it moves to the left, it decreases. Time adjusts the delay time of the Doubler delay lines. Also shown is a visualization for the main LFO, in which its rate, waveform and the phase relationships of the stereo channels are shown.
Also included are Triangle Analog, Triangle 8, Triangle 16, which are described below:. When Phase is chosen, adjusting the Phase value will change the phase relationship between the left and right channels. This can be seen in the LFOs visualization. This is similar to how Pulse Width affects rectangular waveforms, and its effects can be seen in the waveform view of the main LFO. When free-running, the sync frequencies are shown in Hertz and can be adjusted using the LFO2 Freq parameter.
When tempo-synced, beat divisions are shown instead and can be set using the LFO2 Rate parameter. The additional envelope follower and Safe Bass high-pass filter are accessible when the device is unfolded. Amount adjusts the amount of delay modulation that is applied to incoming signals and affects both the main LFO and LFO2. Increasing this sounds more extreme and tends to create a strong comb filtering effect, amplifying some frequencies and attenuating others.
In Doubler Mode, it will also create audible delays if playback is stopped. Caution: high feedback values in combination with certain settings can cause quick increases in volume levels. Be sure to protect your ears and equipment! Below the main LFO you will find an envelope follower. An envelope follower uses the amplitude from an incoming audio signal and translates it to a modulation source.
To use the envelope follower, activate the Env Fol button and set the Envelope Amount value to a value other than zero. Envelope Amount adjusts the intensity of the modulation caused by the envelope follower. Negative values invert the phase of the envelope. Shorter Attack times cause the envelope follower to act more quickly, while longer times delay its onset. Shorter Release times cause the envelope follower to stop its effect faster than longer Release times. Safe Bass is a high-pass filter, effectively reducing the effect on signal components below the specified frequency.
The applicable range is from 5 Hz to Hz. This can make mixing certain bass-heavy material easier.
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