After each gain adjustment, I listen carefully for those first traces of "squashed" sound and I look at the waveform to see how dense it's getting. All my other Mastering effects and plugins are in place and adjusted, so bringing up the loudness is the very last step. Once the track begins to lose its liveliness and sparkle, I turn the gain back down just a little, then export and normalize the track. But again I expect you'll hear a lot more here pretty soon.
Re: Limiter on Master bus Post by Tarekith » Mon Apr 27, pm Personally I recommend people mix without a limiter or compressor, especially if you're not sure how to use one properly. Aim to get the mixdown sounding as good as you can with some headroom I use 6dB, but there's variations on this , then add some limiting to "master" the track yourself. Once you really understand how a compressor works and when to use, then it's ok to use one for color or to glue things a bit, but I definitely don't think it's something you HAVE to have.
Of course only my point of view Post by lonemale » Mon May 25, pm. Post by 7G » Mon May 25, pm. Post by Superchibisan » Mon May 25, pm. Post by Pitch Black » Tue May 26, am. Post by MrShine » Tue May 26, am. Post by drumrak » Tue May 26, am. Post by audiovoid » Sun May 31, am.
Post by michaellpenman » Sun May 31, am. Post by rikstar23 » Fri Jun 26, am. 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 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. Output sets the amount of gain applied to the processed signal. The Warmth control adds slight distortion and filtering for a warmer sound.
The Redux effect has a variety of parameters for creating a wide range of jagged and edgy sounds. You can radically mangle any source material, with effects ranging from harsh distortion and digital aliasing artifacts to warm, fat 8-bit grit. Extra noise and stereo width can be added to the downsampling process, while filtering further transforms the sound. Redux makes use of two different digital signal manipulation techniques: downsampling and bit reduction.
The downsampling controls are available on the left side of the device. The frequency range of the added content is dependent upon the relationship between the frequency content of the material and the sample rate chosen in the device. Rate sets the sample rate to which the signal is degraded.
Lower values result in increased imaging and more inharmonic tones. This results in a noisier sound, as well as increased stereo width. The Filter section has both a Pre and a Post setting. Enabling the Pre button engages a filter before downsampling, which reduces the bandwidth of the signal processed by downsampling.
When Jitter is in use, it also reduces the stereo width of the signal. The Post button engages a low-pass filter after the downsampling process, which reduces imaging. The Post filter frequency can be adjusted using the Post-Filter Octave slider.
The number shown represents the number of octaves above or below half of the frequency shown in the Rate parameter. The bit reduction controls are available on the right side of the device.
Bit reduction decreases the number of bits used to represent the digital signal, reducing dynamic range while adding distortion and noise. At extreme settings, all original dynamics are lost and sounds are reduced to jagged square waves.
The Bits control reduces the number of bits being used. The value shown represents the number of bits used to encode the output signal. Reducing this value increases noise and distortion while reducing dynamic range. Higher values produce a finer resolution for smaller amplitudes, meaning that subtle signal components will be less affected than louder ones.
The total amount of distortion produced with different Shape settings will depend upon the dynamic range of the input signal. Enabling the DC Shift button applies an amplitude offset before the quantization process. This significantly changes the sound of the quantization distortion, especially when Bits is set to lower values, increasing volume and adding crunch!
Bit Reduction is similar, but while downsampling superimposes a grid in time, bit reduction does the same for amplitude. If the Bit Reduction amplitude dial is set to 8, amplitude levels are quantized to steps 8 bit resolution. If set to 1, the result is pretty brutal: Each sample contains either a full positive or full negative signal, with nothing in between.
Bit Reduction defines an input signal of 0dB as 16 bit. Signals above 0dB are clipped, and the red overload LED will light up. This device consists of five parallel resonators that superimpose a tonal character on the input source. It can produce sounds resembling anything from plucked strings to vocoder-like effects. The resonators are tuned in semitones, providing a musical way of adjusting them.
The first resonator defines the root pitch and the four others are tuned relative to this pitch in musical intervals. The input signal passes first through a filter, and then into the resonators. There are four input filter types to select from: lowpass, bandpass, highpass and notch. The input filter frequency can be adjusted with the Frequency parameter. The first resonator is fed with both the left and right input channels, while the second and fourth resonators are dedicated to the left channel, and the third and fifth to the right channel.
The Note parameter defines the root pitch of all the resonators ranging from C-1 to C5. It can also be detuned in cents using the Fine parameter. The Decay parameter lets you adjust the amount of time it takes for the resonators to be silent after getting an input signal.
The longer the decay time, the more tonal the result will be, similar to the behavior of an undamped piano string. As with a real string, the decay time depends on the pitch, so low notes will last longer than higher ones. The Const switch holds the decay time constant regardless of the actual pitch.
Resonators provides two different resonation modes. A resonator that is turned off does not need CPU. Turning off the first resonator does not affect the other ones. Either filter may be switched off when not needed to save CPU power. Predelay controls the delay time, in milliseconds, before the onset of the first early reflection. With small values, the reflections decay more gradually and the diffused sound occurs sooner, leading to a larger overlap between these components.
With large values, the reflections decay more rapidly and the diffused onset occurs later. Spin applies modulation to the early reflections. The X-Y control accesses the depth and frequency of these modulations. A larger depth tends to provide a less-colored more spectrally neutral late diffusion response. If the modulation frequency is too high, doppler frequency shifting of the source sound will occur, along with surreal panning effects.
Spin may be turned off, using the associated switch, for modest CPU savings. The Quality chooser controls the tradeoff between reverb quality and performance. At one extreme, a very large size will lend a shifting, diffused delay effect to the reverb. The other extreme — a very small value — will give it a highly colored, metallic feel. At the highest setting of degrees, each ear receives a reverberant channel that is independent of the other this is also a property of the diffusion in real rooms.
The lowest setting mixes the output signal to mono. The Diffusion network creates the reverberant tail that follows the early reflections. High and low shelving filters provide frequency-dependent reverberation decay. The high-frequency decay models the absorption of sound energy due to air, walls and other materials in the room people, carpeting and so forth. The low shelf provides a thinner decay. Each filter may be turned off to save CPU consumption.
The Freeze control freezes the diffuse response of the input sound. When on, the reverberation will sustain almost endlessly. Cut modifies Freeze by preventing the input signal from adding to the frozen reverberation; when off, the input signal will contribute to the diffused amplitude. Flat bypasses the high and low shelf filters when freeze is on. If Flat is off, the frozen reverberation will lose energy in the attenuated frequency bands, depending on the state of the high and low shelving filters.
The Chorus section adds a little modulation and motion to the diffusion. Like the Spin section, you can control the modulation frequency and amplitude, or turn it off. Saturator is a waveshaping effect that can add that missing dirt, punch or warmth to your sound. It can coat input signals with a soft saturation or drive them into many different flavors of distortion.
The curve defines the transfer function, which is the extent to which output values fluctuate in relation to input values. Because this is usually a nonlinear process, the incoming signal is reshaped to a greater or lesser degree depending upon its level at each moment in time. Incoming signals are first clipped to the dB level set by the Drive control. The meter on the right side of the display shows how much Saturator is influencing the signal.
There is also the flexible Waveshaper mode, featuring six adjustable waveshaping parameters. Secondary option: Saturator as limiter. I swear I used it as such back in Live 4, but now I seem to get noticeable digital clipping whatever I do.
A final request: a freeware clipper for OSX? Thanks for your time. Post by Superchibisan » Mon Oct 20, pm uh, cause you're doing it wrong?
Post by krank » Mon Oct 20, pm Superchibisan wrote: uh, cause you're doing it wrong? Post by Superchibisan » Mon Oct 20, pm because the threshold determines how much of the signal hits the inf. Post by krank » Mon Oct 20, pm Superchibisan wrote: because the threshold determines how much of the signal hits the inf. Post by Superchibisan » Mon Oct 20, pm not seeing hte point here Post by krank » Tue Oct 21, am Superchibisan wrote: not seeing hte point here Post by davez » Wed Oct 22, am krank wrote: Put compressor on master.
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