sfk

 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 

Every effort has been made to ensure that the information in this manual is accurate. Ableton assumes no responsibility or liability for any errors or inaccuracies that may appear in this book. Transforms sound into partials and feeds them into a frequency-based delay, resulting in metallic echoes, frequency-shifted and reverb-like effects.

The Freeze function captures and holds a slice of audio — either free-running or in time with the beat — for stuttered, glitched and washed-out effects. Create jittery glitch effects, delayed digital shimmers and outlandish vibrato with this Max for Live pitch shifting device created in collaboration with Robert Henke and inspired by the Publison DHM 89 — an early digital effects processor.

Six playful instruments and effects created in collaboration with Dillon Bastan that use natural and physical processes as their inspiration. Vector FM — Manipulate different combinations of FM oscillators represented by moving particles — play with movement and voices to create evolving modulations and textures. Vector Grain — A granular looper that visualizes sound modulation by moving particles on the interface — play with attraction and magnetism or loop the particles through a flow field.

Vector Delay — Each visualized particle in this multitap delay device represents a different delay line — create pitch shifting or reverse delay effects with movement controlled by physical forces. Emit — This visual granular synthesizer uses particles shooting across a spectrogram to show the grain of a sample being played, while vertical movements represent filtering and panning. Tree Tone — Grow different fractal patterns inspired by plants, then use them as resonators for internally generated noise or incoming audio.

Bouncy Notes — Bounce balls up and down a piano roll to create pitch shifting delay effects or arpeggiation with this gravity-based MIDI sequencer. A producer and drummer duo show tempo following and Macro improvements in action. Live listens to and adjusts its tempo based on incoming audio in real time, making it a dynamic part of the band instead of the tempo source that everyone has to follow.

Store the state of your Macros for later recall — perfect for creating instant variations to your sounds or builds and drops during performance. Configure your Racks to have between 1 and 16 Macros. Randomize the state of your Macros with the randomization button.

Map this control to MIDI and perform drastic changes in real time to surprise your audience and yourself. Set the probability that a note or drum hit will occur and let Live generate surprising variations to your patterns that change over time.

Define ranges for velocity probability for subtle, humanized variations in the dynamics of your patterns. Follow Actions can now be linked to the clip length, making it faster to create interesting sequences of clips.

Scene Follow Actions let you create evolving arrangements. You can also set Follow Actions to jump to specific clips and enable and disable Follow Actions globally. Upright Piano infuses your productions with the natural warmth of an upright piano. Feature a rich blend of trumpet, flugelhorn, tenor horn and trombone in your music. 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. 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. I still use it to this day. You could spend all of your time here or you can choose to split it between here and the Session View. Session View Tab : The other main view of Live.

If the Arrangement is the linear view of time from start to finish , then Session is the non-linear view of time. This is where you gather, or even create, all of your musical ideas.

You should check out the Browser—it’s all there. Check out the Help View for tutorial and much more. You should probably check this out even if you think you don’t need help. The Clip View allows you to create, view, and edit those ideas. The Mixer will let you adjust the volume, panning, etc. It’s also where the record buttons are—you’ll need them. Now, what is Ableton Live? In a nutshell, it’s a platform for creating and performing music.

 
 

 

Download ableton live 9 windows exe for free.Ableton Reference Manual Version 11 | Ableton

 
It is possible to change the selection of automation and take lane headers using the up and down arrow keys. This overwrites any audio currently stored in Looper. Delay 1 has a highpass filter that can remove low frequencies from the delayed signal. 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.

 
 

Ableton Live for Windows – Download it from Uptodown for free

 
 

Dragging them at one end, while holding down the [Shift] key the old case , will stretch or shrink the audio sample. And the good thing is that you do timestretching at the same time, adapting to the bars you choose, all in sync with the tempo of your project.

This is only one example of what you can do with this new version. Select the MIDI clips you want in the Arrange View, and from their note editor you can arrange and retouch all of them at once.

Another great new feature is the automation process, which is actually a new mode in Ableton Live 10 Suite. When you press the [A] key, the corresponding tabs on the track header appear, as well as the parameter automation lines.

This way you can switch between the Array View and editing in automation mode. Even from the latter, you can still cut, stretch and invert your audio clips. Also now the change points in the automation curves adapt to the bar grid, and if you drag them to the next point, they disappear and reappear if you go backwards. If you also double-click on a blank space in the track, you will create a new automation point.

As in previous versions, holding down the [Alt] key while dragging an automation line will allow you to shape it and create smoother or more abrupt changes.

Simply excellent. We do not miss the obvious reference of the new devices in Ableton Live 10 Suite. Leave a Reply Cancel reply You must be logged in to post a comment. This website uses cookies so that we can provide you with the best user experience possible. Cookie information is stored in your browser and performs functions such as recognising you when you return to our website and helping our team to understand which sections of the website you find most interesting and useful.

Strictly Necessary Cookie should be enabled at all times so that we can save your preferences for cookie settings. If you disable this cookie, we will not be able to save your preferences. This means that every time you visit this website you will need to enable or disable cookies again. Ableton Live Suite Download. The program analyzes your files Stay in the flow with a multitude of workflow improvements.

Max for Live , seamlessly built Kickstart is a program that enables you to get the signature sidechain effect in your own tracks. DVS guitar is a plug-in developed by Dream Vortex studio. It is used with Windows PC sequencing software used It’s an effect. It’s an instrument. It’s Stutter Edit—an innovative tool for both studio and stage, designed by BT and developed by iZotope.

Converting EXE files to Download Minecraft Forge Windows Registry To restore the Windows Windows Mac. Windows Users’ choice Download ableton live 9 windows exe Download ableton live 9 windows exe Most people looking for Ableton live 9 windows exe downloaded: Ableton Live Lite. Ableton Live.