Ableton Live 7 Tips and Tricks: Delaney, Martin: : Books – Download Live 11 manual (PDF)

 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
Live Versions: All Operating System: All You can read the Live manual online. The manual is also included in Live as a PDF. Live 7 English manual. Live is the result of musicians wanting a better way to create, Device Additions and Improvements Download Live 11 manual (PDF).
 
 

Ableton live 7 manual free.Ableton Reference Manual Version 11

 
Apr 14,  · Ableton Live 7 Full Version Free Download. Mar 01, Ableton Live Crack Incl Torrent Free Download Lifetime Ableton Live Crack is a music sequencer with impressive features to amaze both users and the is a digital audio workshop where music is processed to produce the best music tunes and beats. Ableton Live 9. Ableton Reference Manual Version for Windows and Mac. Reference Manual by Michelle Hughes, Sara Riegel, Timothy Beutler, Dennis DeSantis, Ian Gallagher, Kevin Haywood, Rose Knudsen, Gerhard Behles, Jakob Rang, Robert Henke, Torsten Slama. Schönhauser Allee | Berlin, Germany. Download Ableton Live 7 Manual Es PDF. Toggle navigation. Search Login; Register; Home; Ableton Live 7 Manual Es; Download Ableton Live 7 Manual Es PDF for free. Report “Ableton Live 7 Manual Es” Please fill this form, we will try to respond as soon as possible.

 

Welcome to Live — Ableton Reference Manual Version 11 | Ableton.Download Ableton Live Manual Free PDF –

 

Discover our top virtual tours. Amazon Explore Browse now. About the author Follow authors to get new release updates, plus improved recommendations. Martin Delaney. Brief content visible, double tap to read full content. Full content visible, double tap to read brief content. Read more Read less. Customer reviews.

How customer reviews and ratings work Customer Reviews, including Product Star Ratings help customers to learn more about the product and decide whether it is the right product for them. Learn more how customers reviews work on Amazon. Top reviews Most recent Top reviews. Top reviews from the United States.

There was a problem filtering reviews right now. Please try again later. Verified Purchase. This is the book to learn this particular Music Program, by all means, this is not a substitute to the Ableton Manual but merely a companion to that, to get the best results out of this book, you need first to have the actual software install on your computer to follow through with the tips and trick it teaches, basically giving you a basic definition of the feature in question followed by a suggestion example to apply to your musical projects.

The illustrations shown here are quite clear throughout the book with big snapshots showing where you need to be around Ableton at all times which is very helpful, this is the area where most books fall short, drowning the user with endless text with not enough visuals to follow through.

The author Mr. Delaney obviously knows the subject matter very well, with the graphic ability to show not just the basic operations, but additional tricks to apply to your piece of music. Ableton Live 7 Tips and Tricks is really what it is, additional ways to use the program to your advantage beyond the Manual.

I’ve been using Ableton Live since version 5 for Multimedia projects and even I’ve learn different ways of applying effects, dynamics and using hardware keyboards and controllers and better implementing Ableton Live with different Audio and Midi interfaces along with other Software’s as VST or Rewire Conclusion: To get the best results out of this book, I suggest reading the Manual first and the built-in Tutorials that comes with Ableton Live to get real world satisfying results out of your Multimedia.

Newcomers, Intermediate and Advanced User will all benefit greatly from this book. There are certainly good tips in this book and it manages to avoid being just a re-hash of the Ableton Live user’s manual.

It’s greatest problem is an almost total lack of information on certain fundamentals that make Ableton Live the product of choice for so many DJ’s and producers. An example is time Warping, which barely gets a mention in the book, and has been an important part of Live for so long that it’s nearly unforgivable to give it so little attention.

Still, for the price, there are enough tidbits in this book that you won’t find in the Ableton user’s guide to make it worth the purchase. A nice read with plenty of pictures supporting the text. If you can get it cheap do it, you won’t regret.

Nice gift for professionals. I bought this book hoping it would kickstart my working with Live. It really hasn’t. The book claims to be something different than the manual. Well, I’m not so sure. Since it wants to cover all aspects of Live it tends to be pretty shallow. Maybe I’m unfair, but to me it reads a bit like a salespersons description of the software.

There’s even a whole chapter devoted to midi-controllers, that also seem very much like salestalk for Behringer, M-audio etc. What I would have like would have been regular lessons – tutorials guiding you through step after step. I don’t recommend this book. New Live-users would do a lot better at looking at all the free resources at the Ableton website. There you’ll find great video tutorials as well as good lessons on pdf-files that you could print or read on screen.

While it might not be relevant for Live, but just to show I’m not impossible to please: I bought a great book for another software – Propellerhead’s Reason 4. The book “Reason 4 Ignite” by Matt Piper has a great structure that walks you through different parts of the software in well written examples. Around pages of content reside in this book and it’s broken up over 21 chapters.

Only complaint is that this book should have been published in color. The black and white pictures don’t make for a great experience and this was a mistake but the people who bankrolled this book. The Gain knob adjusts the audio level coming back from the sound source.

This level should be set carefully to avoid clipping. Since external devices can 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.

Any latency introduced by devices within Live will be compensated for automatically, so the slider will be disabled when using the External Instrument Device to route internally. Note: If the Delay Compensation option see Impulse is a drum sampler with complex modulation capabilities. Alternatively, each sample slot features a Hot-Swap button for hot-swapping samples see 5.

Imported samples are automatically mapped onto your MIDI keyboard, providing that it is plugged in and acknowledged by Live. C3 on the keyboard will trigger the leftmost sample, and the other samples will follow suit in the octave from C3 to C4.

Mapping can be transposed from the default by applying a Pitch device see Each of the eight samples has a proprietary set of parameters, located in the area below the sample slots and visible when the sample is clicked. Adjustments to sample settings are only captured once you hit a new note — they do not affect currently playing notes.

Note that this behavior also defines how Impulse reacts to parameter changes from clip envelopes or automation, which are applied once a new note starts. If you want to achieve continuous changes as a note plays, you may want to use the Simpler see This was designed with a specific situation in mind but can, of course, be used for other purposes : Replicating the way that closed hi-hats will silence open hi-hats.

Each slot can be played, soloed, muted or hot-swapped using controls that appear when the mouse hovers over it. The Start control defines where Impulse begins playing a sample, and can be set up to ms later than the actual sample beginning.

The Stretch control has values from to percent. Negative values will shorten the sample, and positive values will stretch it. Two different stretching algorithms are available: Mode A is ideal for low sounds, such as toms or bass, while Mode B is better for high sounds, such as cymbals. The Filter section offers a broad range of filter types, each of which can impart different sonic characteristics onto the sample by removing certain frequencies.

The Frequency control defines where in the harmonic spectrum the filter is applied; the Resonance control boosts frequencies near that point.

The Saturator gives the sample a fatter, rounder, more analog sound, and can be switched on and off as desired. The Drive control boosts the signal and adds distortion. Extreme Drive settings on low-pitched sounds will produce the typical, overdriven analog synth drum sounds. The envelope can be adjusted using the Decay control, which can be set to a maximum of Impulse has two decay modes: Trigger Mode allows the sample to decay with the note; Gate Mode forces the envelope to wait for a note off message before beginning the decay.

This mode is useful in situations where you need variable decay lengths, as is the case with hi-hat cymbal sounds. Each sample has Volume and Pan controls that adjust amplitude and stereo positioning, respectively. Both controls can be modulated: Pan by velocity and a random value, and Volume by velocity only. Volume adjusts the overall level of the instrument, and Transp adjusts the transposition of all samples. The Time control governs the time-stretching and decay of all samples, allowing you to morph between short and stretched drum sounds.

When a new instance of Impulse is dragged into a track, its signal will be mixed with those of the other instruments and effects feeding the audio chain of the track. It can oftentimes make more sense to isolate the instrument or one of its individual drum samples, and send this signal to a separate track.

Please see the Routing chapter see Operator includes a filter section, an LFO and global controls, as well as individual envelopes for the oscillators, filter, LFO and pitch. The interface of Operator consists of two parts: the display surrounded on either side by the shell.

The shell offers the most important parameters in a single view and is divided into eight sections. On the left side, you will find four oscillator sections, and on the right side from top to bottom, the LFO, the filter section, the pitch section and the global parameters. If you change one of the shell parameters, the display in the center will automatically show the details of the relevant section.

Operator can be folded with the triangular button at its upper left. This is convenient if you do not need to access the display details. Operator offers eleven predefined algorithms that determine how the oscillators are connected. An algorithm is chosen by clicking on one of the structure icons in the global display, which will appear if the bottom right global section of the shell is selected. Signals will flow from top to bottom between the oscillators shown in an algorithm icon.

The algorithm selector can be mapped to a MIDI controller, automated, or modulated in real time, just like any other parameter. Typically, FM synthesis makes use of pure sine waves, creating more complex waveforms via modulation. However, in order to simplify sound design and to create a wider range of possible sounds, we designed Operator to produce a variety of other waveforms, including two types of noise.

You can also draw your own waveforms via a partial editor. The instrument is made complete with an LFO, a pitch envelope and a filter section. Operator will keep you busy if you want to dive deep into sound design! The oscillators come with a built-in collection of basic waveform types — sine, sawtooth, square, triangle and noise — which are selected from the Wave chooser in the individual oscillator displays.

The first of these waveforms is a pure, mathematical sine wave, which is usually the first choice for many FM timbres. The square, triangle and sawtooth waveforms are resynthesized approximations of the ideal shape. The numbers included in the displayed name e. Lower numbers sound mellower and are less likely to create aliasing when used on high pitches. There are also two built-in noise waveforms. You can also select one of the built-in waveforms and then edit it in the same way. The small display next to the Wave chooser gives a realtime overview of your waveform.

When your mouse is over the Oscillator display area, the cursor will change to a pencil. Drawing in the display area then raises or lowers the amplitudes of the harmonics. Holding Shift and dragging will constrain horizontal mouse movement, allowing you to adjust the amplitude of only one harmonic at a time. You can switch between editing the first 16, 32 or 64 harmonics via the switches to the right of the display. Higher harmonics can be generated by repeating the drawn partials with a gradual fadeout, based on the settings in the Repeat chooser.

Low Repeat values result in a brighter sound, while higher values result in more high-end roll-off and a more prominent fundamental. With Repeat off, partials above the 16th, 32nd or 64th harmonic are truncated. The context menu also offers an option to toggle Normalize on or off. When disabled, additional harmonics add additional level.

Note that the volume can become extremely loud if Normalize is off. You can export your waveform in. Ams files can also be loaded into Simpler or Sampler. The frequency of an oscillator can be adjusted in the shell with its Coarse and Fine controls. This can be done for each individual oscillator by activating the Fixed option.

This allows the creation of sounds in which only the timbre will vary when different notes are played, but the tuning will stay the same. Fixed Mode would be useful, for example, in creating live drum sounds. Fixed Mode also allows producing very low frequencies down to 0.

Note that when Fixed Mode is active, the frequency of the oscillator is controlled in the shell with the Frequency Freq and Multiplier Multi controls. This feature can be very useful when working with sequenced sounds in which the velocity of each note can be adjusted carefully. Part of this functionality is the adjacent Q Quantize button.

If this control is activated, the frequency will only move in whole numbers, just as if the Coarse control were being manually adjusted. If quantize is not activated, the frequency will be shifted in an unquantized manner, leading to detuned or inharmonic sounds which very well could be exactly what you want The amplitude of an oscillator depends on the Level setting of the oscillator in the shell and on its envelope, which is shown and edited when the Envelope display is visible.

The phase of each oscillator can be adjusted using the Phase control in its display. With the R Retrigger button enabled, the waveform restarts at the same position in its phase each time a note is triggered. With R disabled, the oscillator is free-running. When an oscillator is modulating another oscillator, two main properties define the result: the amplitude of the modulating oscillator and the frequency ratio between both oscillators.

Any oscillator that is not modulated by another oscillator can modulate itself, via the Feedback parameter in its display. Aliasing distortion is a common side effect of all digital synthesis and is the result of the finite sample rate and precision of digital systems.

It mostly occurs at high frequencies. FM synthesis is especially likely to produce this kind of effect, since one can easily create sounds with lots of high harmonics. Aliasing is a two-fold beast: A bit of it can be exactly what is needed to create a cool sound, yet a bit too much can make the timbre unplayable, as the perception of pitch is lost when high notes suddenly fold back into arbitrary pitches.

Operator minimizes aliasing by working in a high-quality Antialias mode. This is on by default for new patches, but can be turned off in the global section. The Tone parameter in the global section also allows for controlling aliasing.

Its effect is sometimes similar to a lowpass filter, but this depends on the nature of the sound itself and cannot generally be predicted. If you want to familiarize yourself with the sound of aliasing, turn Tone up fully and play a few very high notes.

You will most likely notice that some notes sound completely different from other notes. Now, turn Tone down and the effect will be reduced, but the sound will be less bright. The LFO in Operator can practically be thought of as a fifth oscillator. It runs at audio rates, and it modulates the frequency of the other oscillators.

It is possible to switch LFO modulation on or off for each individual oscillator and the filter using the Dest. A slider. The LFO can also be turned off entirely if it is unused. The Dest. B chooser allows the LFO to modulate an additional parameter. The intensity of this modulation is determined by the Dest. B slider. Sample and hold uses random numbers chosen at the rate of the LFO, creating the random steps useful for typical retro-futuristic sci-fi sounds.

The noise waveform is simply bandpass-filtered noise. Tip: FM synthesis can be used to create fantastic percussion sounds, and using the LFO with the noise waveform is the key to great hi-hats and snares. The frequency of the LFO can follow note pitch, be fixed or be set to something in between. With the R Retrigger button enabled, the LFO restarts at the same position in its phase each time a note is triggered. With R disabled, the LFO is free-running. This parameter scales both the Dest.

Operator has seven envelopes: one for each oscillator, a filter envelope, a pitch envelope and an envelope for the LFO. All envelopes feature some special looping modes. Additionally, the filter and pitch envelopes have adjustable slopes. A rate is the time it takes to go from one level to the next. As mentioned above, the filter and pitch envelopes also have adjustable slopes.

Clicking on the diamonds between the breakpoints allows you to adjust the slope of the envelope segments. Positive slope values cause the envelope to move quickly at the beginning, then slower. Negative slope values cause the envelope to remain flat for longer, then move faster at the end. A slope of zero is linear; the envelope will move at the same rate throughout the segment.

With FM synthesis, it is possible to create spectacular, endless, permuting sounds; the key to doing this is looping envelopes. Loop Mode can be activated in the lower left corner of the display. If an envelope in Operator is in Loop Mode and reaches sustain level while the note is still being held, it will be retriggered. The rate for this movement is defined by the Loop Time parameter. Note that envelopes in Loop Mode can loop very quickly and can therefore be used to achieve effects that one would not normally expect from an envelope generator.

While Loop Mode is good for textures and experimental sounds, Operator also includes Beat and Sync Modes, which provide a simple way of creating rhythmical sounds. If set to Beat Mode, an envelope will restart after the beat time selected from the Repeat chooser. In Beat Mode, the repeat time is defined in fractions of song time, but notes are not quantized. If you play a note a bit out of sync, it will repeat perfectly but stay out of sync.

In Sync Mode however, the first repetition is quantized to the nearest 16th note and, as a result, all following repetitions are synced to the song tempo. Note that Sync Mode only works if the song is playing, and otherwise it will behave like Beat Mode. Note: To avoid the audible clicks caused by restarting from its initial level, a looped envelope will restart from its actual level and move with the set attack rate to peak level. There is also a mode called Trigger that is ideal for working with percussive sounds.

In this mode, note off is ignored. This means that the length of time a key is held has no effect on the length of the sound. The rates of all the envelopes in Operator can be scaled in unison by the Time control in the global section of the shell.

Note that beat-time values in Beat and Sync Modes are not influenced by the global Time parameter. These modulations in conjunction with the loop feature can be used to create very, very complex things A slider and the envelope can be turned off altogether via the switch in the pitch section of the shell. Like the LFO, the pitch envelope can modulate an additional parameter as chosen by the Dest.

B chooser. The intensity of this modulation is determined by the Amt. B slider and the main Pitch Env value. The pitch and filter envelopes each have an additional parameter called End, which determines the level the envelope will move to after the key is released. The rate of this envelope segment is determined by the release time. And, since the envelope of the LFO itself can loop, it can serve as a third LFO modulating the intensity of the first! And, since the oscillators also provide you with the classic waveforms of analog synthesizers, you can very easily build a subtractive synthesizer with them.

Operator offers a variety of filter types including lowpass, highpass, bandpass, notch, and a special Morph filter. Each filter can be switched between 12 and 24 dB slopes as well as a selection of analog-modeled circuit behaviors developed in conjunction with Cytomic that emulate hardware filters found on some classic analog synthesizers. This is available for all of the filter types. The OSR circuit option is a state-variable type with resonance limited by a unique hard-clipping diode.

This is modeled on the filters used in a somewhat rare British monosynth, and is available for all filter types. The MS2 circuit option uses a Sallen-Key design and soft clipping to limit resonance. It is modeled on the filters used in a famous semi-modular Japanese monosynth and is available for the lowpass and highpass filters. The SMP circuit is a custom design not based on any particular hardware. It shares characteristics of both the MS2 and PRD circuits and is available for the lowpass and highpass filters.

The PRD circuit uses a ladder design and has no explicit resonance limiting. It is modeled on the filters used in a legacy dual-oscillator monosynth from the United States and is available for the lowpass and highpass filters. The most important filter parameters are the typical synth controls Frequency and Resonance.

Frequency determines where in the harmonic spectrum the filter is applied; Resonance boosts frequencies near that point.

When using the lowpass, highpass, or bandpass filter with any circuit type besides Clean, there is an additional Drive control that can be used to add gain or distortion to the signal before it enters the filter.

The Morph filter has an additional Morph control which sweeps the filter type continuously from lowpass to bandpass to highpass to notch and back to lowpass. Filter cutoff frequency and resonance can be adjusted in the shell or by dragging the filter response curve in the display area.

Filter frequency can also be modulated by the following:. The Shaper Drive Shp. If you open a Set that was created in a version of Live older than version 9. These consist of 12 dB or 24 dB lowpass, bandpass and highpass filters, as well as a notch filter, and do not feature a Drive control.

Each Operator loaded with the legacy filters shows an Upgrade button in the title bar. Pressing this button will permanently switch the filter selection to the newer models for that instance of Operator. Note that this change may make your Set sound different. Additionally, the global display area provides a comprehensive set of modulation routing controls.

The maximum number of Operator voices notes playing simultaneously can be adjusted with the Voices parameter in the global display. Ideally, one would want to leave this setting high enough so that no voices would be turned off while playing, however a setting between 6 and 12 is usually more realistic when considering CPU power.

Tip: Some sounds should play monophonically by nature, which means that they should only use a single voice. A flute is a good example. In these cases, you can set Voices to 1. If Voices is set to 1, another effect occurs: Overlapping voices will be played legato, which means that the envelopes will not be retriggered from voice to voice, and only pitch will change.

The center of the global display allows for a wide variety of internal MIDI mappings. For more information about the available modulation options, see the complete parameter list see Operator includes a polyphonic glide function.

When this function is activated, new notes will start with the pitch of the last note played and then slide gradually to their own played pitch. Glide can be turned on or off and adjusted with the Glide Time control in the pitch display. Operator also offers a special Spread parameter that creates a rich stereo chorus by using two voices per note and panning one to the left and one to the right. The two voices are detuned, and the amount of detuning can be adjusted with the Spread control in the pitch section of the shell.

Tip: Whether or not spread is applied to a particular note depends upon the setting of the Spread parameter during the note-on event. To achieve special effects, you could, for instance, create a sequence where Spread is 0 most of the time and turned on only for some notes. These notes will then play in stereo, while the others will play mono. Note: Spread is a CPU-intensive parameter. If you want to save CPU power, turn off features that you do not need or reduce the number of voices.

For the sake of saving CPU resources, you will also usually want to reduce the number of voices to something between 6 and 12, and carefully use the Spread feature. The Interpolation and Antialias modes in the global display can also be turned off to conserve CPU resources. FM synthesis was first explored musically by the composer and computer music pioneer John Chowning in the mids. In , he and Stanford University began a relationship with Yamaha that lead to one of the most successful commercial musical instruments ever, the DX7.

John Chowning realized some very amazing and beautiful musical pieces based on a synthesis concept that you can now explore yourself simply by playing with Operator in Live. The function of each Operator parameter is explained in the forthcoming sections. Remember that you can also access explanations of controls in Live including those belonging to Operator directly from the software by placing the mouse over the control and reading the text that appears in the Info View. Parameters in this list are grouped into sections based on where they appear in Operator.

Tone — Operator is capable of producing timbres with very high frequencies, which can sometimes lead to aliasing artifacts. The Tone setting controls the high frequency content of sounds. Higher settings are typically brighter but also more likely to produce aliasing. Algorithm — An oscillator can modulate other oscillators, be modulated by other oscillators, or both. The algorithm defines the connections between the oscillators and therefore has a significant impact on the sound that is created.

Voices — This sets the maximum number of notes that can sound simultaneously. If more notes than available voices are requested, the oldest notes will be cut off. Retrigger R — When enabled, notes that are enabled will be retriggered, rather than generating an additional voice. Interpolation — This toggles the interpolation algorithm of the oscillators and the LFO. If turned off, some timbres will sound more rough, especially the noise waveform. Turning this off will also save some CPU power.

Disabling this modes reduces the CPU load. Pan — Use this to adjust the panorama of each note. This is especially useful when modulated with clip envelopes.

Typically this is used for piano-like sounds. These modulation targets are available as MIDI routing destinations in the global display, and also as modulation targets for the LFO and pitch envelope. OSC Feedback — Modulates the amount of feedback for all oscillators. Note that feedback is only applied to oscillators that are not modulated by other oscillators.

FM Drive — Modulates the volume of all oscillators which are modulating other oscillators, thus changing the timbre. Filter Drive — Modulates the amount of the Drive not available when the Morph filter is selected. Pitch Envelope On — This turns the pitch envelope on and off. Turning it off if it is unused saves some CPU power.

Spread — If Spread is turned up, the synthesizer uses two detuned voices per note, one each on the left and right stereo channels, to create chorusing sounds.

Spread is a very CPU-intensive effect. Transpose — This is the global transposition setting for the instrument. Changing this parameter will affect notes that are already playing.

It is therefore listed in the section on envelopes see Glide G — With Glide on, notes will slide from the pitch of the last played note to their played pitch. Note that all envelopes are not retriggered in this case if notes are being played legato. Glide Time Time — This is the time it takes for a note to slide from the pitch of the last played note to its final pitch when Glide is activated.

This setting has no effect if Glide is not activated. Pitch Envelope to Osc Destination A-D — The pitch envelope affects the frequency of the respective oscillator if this is turned on. Pitch Envelope Destination B — This sets the second modulation destination for the pitch envelope. Filter On — This turns the filter on and off.

Turning it off when it is unused saves CPU power. Filter Type — This chooser selects from lowpass, highpass, bandpass, notch, and Morph filters. Circuit Type — This chooser selects from a variety of circuit types that emulate the character of classic analog synthesizers.

Filter Frequency Freq — This defines the center or cutoff frequency of the filter. Note that the resulting frequency may also be modulated by note velocity and by the filter envelope.

Filter Resonance Res — This defines the resonance around the filter frequency of the lowpass and highpass filters, and the width of the bandpass and notch filters. The center point for this function is C3. It is therefore listed in the section on envelopes. Filter Drive Flt. Drive — Applies additional input gain to the signal before it enters the filter. Shaper Drive Shp. Drive — This boosts or attenuates the signal level being sent to the waveshaper.

Turning it off when it is unused saves some CPU power. All waveforms are band-limited to avoid unwanted clicks. Due to the possible high frequencies, the LFO can also function as a fifth oscillator. Retrigger R — When enabled, the LFO restarts at the same position in its phase each time a note is triggered. Note that the actual effect also depends on the LFO envelope.

Osc Coarse Frequency Coarse — The relationship between oscillator frequency and note pitch is defined by the Coarse and Fine parameters. Coarse sets the ratio in whole numbers, creating a harmonic relationship. Osc Fine Frequency Fine — The relationship between oscillator frequency and note pitch is defined by the Coarse and Fine parameters. Fine sets the ratio in fractions of whole numbers, creating an inharmonic relationship.

This frequency is constant, regardless of note pitch. Osc Fixed Multiplier Multi — This is used to adjust the range of the fixed frequency. Osc Output Level Level — This sets the output level of the oscillator.

If this oscillator is modulating another, its level has significant influence on the resulting timbre. Osc Waveform Wave — Choose from a collection of carefully selected waveforms. You can then edit them via the harmonics editor. Osc Feedback Feedback — An oscillator can modulate itself if it is not modulated by another oscillator. The modulation is dependent not only on the setting of the feedback control but also on the oscillator level and the envelope.

Higher feedback creates a more complex resulting waveform. Osc Phase Phase — This sets the initial phase of the oscillator. The range represents one whole cycle. Retrigger R — When enabled, the oscillator restarts at the same position in its phase each time a note is triggered.

Repeat — Higher harmonics can be generated by repeating the drawn partials with a gradual fadeout, based on the settings in the Repeat chooser. If activated, the sonic result is the same as manually changing the Coarse parameter for each note.

Applying this to modulating oscillators creates velocity-dependent timbres. Envelope Attack Time Attack — This sets the time it takes for a note to reach the peak level, starting from the initial level. For the oscillator envelopes, the shape of this segment of the envelope is linear. For the filter and pitch envelopes, the shape of the segment can be adjusted. Envelope Decay Time Decay — This sets the time it takes for a note to reach the sustain level from the peak level.

For the oscillator envelopes, the shape of this segment of the envelope is exponential. Envelope Release Time Release — This is the time it takes for a note to reach the end level after a note-off message is received. For the oscillator envelopes, this level is always -inf dB and the shape of the segment is exponential. For the filter and pitch envelopes, the end level is determined by the End Level parameter and the shape of the segment can be adjusted.

This envelope segment will begin at the value of the envelope at the moment the note-off message occurs, regardless of which segment is currently active. Envelope Sustain Level Sustain — This is the sustain level at the end of the note decay. The envelope will stay at this level until note release unless it is in Loop, Sync or Beat Mode. Envelope Loop Mode Loop — If this is set to Loop, the envelope will start again after the end of the decay segment.

If set to Beat or Sync, it will start again after a given beat-time. In Sync Mode, this behavior will be quantized to song time. In Trigger mode, the envelope ignores note off. When retriggered, the envelope will move at the given attack rate from the current level to the peak level. The time it takes to move from the sustain level to the initial value is defined by this parameter. This is especially interesting if the envelopes are looping.

Note that this modulation does not influence the beat-time in Beat or Sync Modes, but the envelope segments themselves. The filter and pitch envelopes also provide parameters that adjust the slope of their envelope segments. These include:. There is also a command to export the waveform as an.

When enabled, E3 is the center. When disabled, C3 is the center. Note that this option is only available when loading Operator presets that were made in versions of Live prior to Live 9. Sampler users who want to share their presets with all Live users can convert their work to Simpler see It has been designed from the start to handle multi-gigabyte instrument libraries with ease, and it imports most common library formats.

Getting started with Sampler is as easy as choosing a preset from the browser. Presets imported from third-party sample libraries are listed here, too, in the Imports folder. Once you have loaded a Sampler preset into a track, remember to arm the track for recording which also enables you to hear any MIDI notes you might want to play , and then start playing!

This technique is used to accurately capture the complexity of instruments that produce dynamic timbral changes. Rather than rely on the simple transposition of a single recorded sample, multisampling captures an instrument at multiple points within its critical sonic range. This typically means capturing the instrument at different pitches as well as different levels of emphasis played softly, moderately, loudly, etc.

The resulting multisample is a collection of all the individually recorded sample files. The acoustic piano, for example, is a commonly multisampled instrument.

Sampler is designed to let you approach multisampling on whatever level you like: you can load and play multisample presets, import multisamples from third-party vendors see Lastly, you do not have use multisamples at all — drop a single sample into Sampler and take advantage of its internal modulation system however you like. Fold — Folds Sampler so that only the device title bar is visible. Unfold quickly by double-clicking the device title bar. Show Preset Name — By default, Sampler takes the top-most sample in the sample layer list as its title.

By default, Sampler will automatically be locked to the control surface when the track is armed for recording. A hand icon in the title bar of locked devices serves as a reminder of their statuses. Uncheck this to enable linear crossfades at looping points. Clicking a tab will, with the exception of the Zone tab, reveal its properties below.

In addition to serving as an organizational aid, each tab has one or more LEDs that indicate if there is modulation information in the corresponding area. We will get to know Sampler by examining each of these tabs. The Zone Editor opens in its own dedicated view, directly above the Device View.

On the left side of the Zone Editor is the sample layer list, where multisamples are organized. All of the individual samples belonging to a multisample are shown in this list, where they are referred to as layers. For complex multisamples, this list can be quite long. The rest of the view is occupied by one of three editors that correspond to the sample layers: the Key Zone Editor see With Auto Select enabled, all sample layers that are able to play an incoming note will become selected in the sample layer list for the duration of that note.

All samples contained in the currently loaded multisample are listed here, with each sample given its own layer. For very large multisamples, this list might be hundreds of layers long!

Fortunately, layers can be descriptively named according to their root key, for example. Mousing over a layer in the list or a zone in the zone editors will display relevant information about the corresponding sample in the Status Bar bottom of screen. Selecting any layer will load its sample into the Sample tab for examination. Distribute Ranges Around Root Key — For layers that have different root keys, this option will distribute their ranges as evenly as possible around their root keys, but without overlapping.

For layers that share a root key, the ranges will be distributed evenly. Note that this does not necessarily return panned stereo samples to the center position; rather, Live intelligently calculates a pan position for an even stereo spread. Select All With Same Range — Selects all layers whose zone range matches the currently selected layer. Sort Alphabetically Ascending and Descending — Arranges samples alphabetically according to their names.

Sort by Key Ascending and Descending — Sorts key zones in an ascending or descending pattern. Sort by Velocity Ascending and Descending — Sorts velocity zones in an ascending or descending pattern. Sort by Selector Ascending and Descending — Sorts sample select zones in an ascending or descending pattern. Key zones define the range of MIDI notes over which each sample will play. Samples are only triggered when incoming MIDI notes lie within their key zone.

Every sample has its own key zone, which can span anywhere from a single key up to the full A typical multisampled instrument contains many individual samples, distributed into many key zones. By default, the key zones of newly imported samples cover the full MIDI note range. Zones can be moved and resized like clips in the Arrangement View, by dragging their right or left edges to resize them, then dragging them into position. Zones can also be faded over a number of semitones at either end by dragging their top right or left corners.

This makes it easy to smoothly crossfade between adjacent samples as the length of the keyboard is traversed. The Lin and Pow boxes above the sample layer list indicate whether the zones will fade in a linear or exponential manner. The timbre of most musical instruments changes greatly with playing intensity. Therefore, the best multisamples capture not only individual notes, but also each of those notes at different velocities. The Velocity Zone Editor, when toggled, appears alongside the sample layer list.

Velocity is measured on a scale of , and this number range appears across the top of the editor. Each sample also has a Sample Select zone, which is a data filter that is not tied to any particular kind of MIDI input. Sample Select zones are very similar to the Chain Select Zones see The Sample Select Editor, when toggled, appears alongside the sample layer list. The editor has a scale of , similar to the Velocity Zone Editor.

Above the value scale is the draggable indicator known as the sample selector. Please note that the position of the sample selector only determines which samples are available for triggering. Once a sample has been triggered, changing the position of the sample selector will not switch to a different sample during playback.

The playback behavior of individual samples is set within the Sample tab. Most of this tab is dedicated to displaying the waveform of the currently selected sample. Hovering your mouse over the waveform will display relevant information about the sample in the Status Bar bottom of screen. It is important to keep in mind that most of the values in this tab reflect the state of the currently selected sample only. Reverse — This is a global, modulatable control that reverses playback of the entire multisample.

Unlike the Reverse function in the Clip View, a new sample file is not generated. Instead, sample playback begins from the Sample End point, proceeds backwards through the Sustain Loop if active , and arrives at the Sample Start point. Snap — Snaps all start and end points to the waveform zero-crossings points where the amplitude is zero to avoid clicks. You can quickly see this by using Snap on square wave samples.

As with Simpler, this snap is based on the left channel of stereo samples, so a small Crossfade value may be necessary in some cases to completely eliminate clicks. Sample — Displays the name of the current sample layer, and can be used to quickly select different layers of the loaded multisample.

Envelope Attack Time Attack — This sets the time it takes for an envelope to reach the peak level, starting from the initial level. The shape of the attack can be adjusted via the Attack Slope A. Slope parameter. Envelope Decay Time Decay — This sets the time it takes for an envelope to reach the sustain level from the peak level. The shape of the decay can be adjusted via the Decay Slope D. Envelope Sustain Level Sustain — This is the sustain level at the end of the envelope decay.

Envelope Release Time Release — This is the time it takes for an envelope to reach the end level after a note-off message is received. The shape of this stage of the envelope is determined by the Release Slope R. Slope value. Envelope Peak Level Peak — This is the peak level at the end of the envelope attack, and the beginning of the Decay stage. Sample Start — The time value where playback will begin. Sample End — The time value where playback will end unless a loop is enabled , even if the volume envelope has not ended.

Sustain Mode — The optional Sustain Loop defines a region of the sample where playback will be repeated while the note is in the sustain stage of its envelope. Activating the Sustain Loop also allows the Release Loop to be enabled. This creates several playback options:. No Sustain Loop — Playback proceeds linearly until either the Sample End is reached or the volume envelope completes its release stage. Sustain Loop Enabled — Playback proceeds linearly until Loop End is reached, when it jumps immediately to Loop Start and continues looping.

If Release Mode is OFF, this pattern continues until the volume envelope has completed its release stage. Release Enabled — When the volume envelope reaches its release stage, playback will proceed linearly towards Sample End. Release Loop Enabled — When the volume envelope reaches its release stage, playback will proceed linearly until reaching Sample End, where it jumps immediately to Release Loop and continues looping until the volume envelope has completed its release stage.

Back-and-Forth Release Loop Enabled — When the volume envelope reaches its release stage, playback will proceed linearly until reaching Sample End, then reverses until it reaches Release Loop, then proceeds again towards Sample End. This pattern continues until the volume envelope has completed its release stage. Release Loop — sets the start position of the Release Loop. The end of the Release Loop is the Sample End.

Sustain- and Release-Loop Crossfade Crossfade — Loop crossfades help remove clicks from loop transitions. By default, Sampler uses constant-power fades at loop boundaries. Tip: this is especially noticeable with very short loops. With Detune, the pitch of these regions can be matched to the rest of the sample. Interpolation Interpol — This is a global setting that determines the accuracy of transposed samples.

This mode can give better performance when modulating start and end points, but loading large multisamples into RAM will quickly leave your computer short of RAM for other tasks.

In any case, it is always recommended to have as much RAM in your computer as possible, as this can provide significant performance gains. Additionally, you can zoom in or out of playing or looping regions, depending on which Sustain and Loop Modes are selected. Vertical Zoom slider — Magnifies the waveform height in the sample display. This is for visual clarity only, and does not affect the audio in any way.

Sampler features one dedicated modulation oscillator per voice, which can perform frequency or amplitude modulation FM or AM on the multisample. The oscillator is fully featured, with 21 waveforms available in the Type chooser , plus its own loopable amplitude envelope for dynamic waveshaping. Note that this oscillator performs modulation only — its output is never heard directly.

What you hear is the effect of its output upon the multisample. FM — In this mode, the modulation oscillator will modulate the frequency of samples, resulting in more complex and different-sounding waveforms. AM — In this mode, the modulation oscillator will modulate the amplitude of samples. Subsonic modulator frequencies result in slow or rapid variation in the volume level; audible modulator frequencies result in composite waveforms.

For detailed information on how these work, see the Sample Playback section see Additionally, the right side of the modulation oscillator section features the following controls:. This determines the depth of the modulation. This is only available when Fixed is set to Off. The pitch envelope modulates the pitch of the sample over time, as well as of the Modulation Oscillator, if it is enabled. On the lower-left of the Pitch Envelope section is the Amount slider.

The actual range depends upon the dynamics of the envelope itself. Spread — When Spread is used, two detuned voices are generated per note. This also doubles the processing requirements.

Good for getting interesting artifacts from multisamples. Glide — The global Glide mode, used in conjunction with the Time parameter to smoothly transition between pitches. Time — Enabling a Glide mode produces a smooth transition between the pitch of played notes.

This parameter determines the length of the transition. Sampler features a polyphonic filter with an optional integrated waveshaper. The filter section offers a variety of filter types including lowpass, highpass, bandpass, notch, and a special Morph filter.

Env button. Below the Filter is a waveshaper, which is toggled by clicking the Shaper button. Four different curves can be chosen for the waveshaper in the Type selector: Soft, Hard, Sine and 4bit. In addition, the signal flow direction can be adjusted with the button above the waveshaper area: with the triangle pointing up, the signal passes from the shaper to the filter; with the triangle pointing down, it passes from the filter to the shaper.

Each Sampler loaded with the legacy filters shows an Upgrade button in the title bar. Pressing this button will permanently switch the filter selection to the newer models for that instance of Sampler. Please see the Sample Playback section see This envelope can also be looped via the Loop chooser. For Loop and Trigger modes, if a note is still held when the Decay stage ends, the envelope will restart from its initial value.

The time it takes to move from the Sustain level to the initial value is defined by the Time parameter. For Beat and Sync modes, if a note is still held after the amount set in the Repeat slider, the envelope will restart from its initial value. Time Global Time Envelope will proportionally shrink or expand the length of all envelopes in Sampler. Finally, the Voices selector provides up to 32 simultaneous voices for each instance of Sampler. Voice retriggering can optionally be enabled by activating the Retrigger button R to the right of the Voices chooser.

When activated, notes which are already playing will be retriggered, rather than generating an additional voice. Turning Retrigger on can save CPU power, especially if a note with a long release time is being triggered very often and very quickly. The Modulation tab offers an additional loopable envelope, plus three LFOs, all capable of modulating multiple parameters, including themselves.

This envelope can be routed to 29 destinations in both the A and B choosers. How much the Auxiliary envelope will modulate destinations A and B is set in the two sliders to the right.

With the note head selected, the LFO will be synced to beat-time, adjustable in the Beats slider to the right. Use this, for example, to gradually introduce vibrato as a note is held.

Volume Vol — LFO 1 can modulate the global volume level. This slider determines the depth of the modulation on a scale.

Pan Pan — LFO 1 can modulate the global pan position. Pitch — LFO 1 can modulate the pitch of samples.

In phase mode, the right and left LFO channels run at equal speed, and the Phase parameter is used to offset the right channel from the left. At the bottom is a Pitch Bend Range slider 0 to 24 steps. The bit range of pitch wheel values can be scaled to produce up to 24 semitones of pitch bend in Sampler.

Finally, clicking in the Sampler image on the right will trigger a scrolling, movie-like credits for Sampler. These are the people you can thank! Sampler can use multisamples created by a number of other software and hardware samplers. This will import it into your User Library. Note that some multisample files will be converted to Instrument Rack see Chapter 20 presets that contain several Sampler instances used to emulate the original more accurately.

This means the new Sampler presets will work regardless of whether the original multisample file is still available. Note: Kontakt. Simpler is an instrument that integrates the basic elements of a sampler with a set of classic synthesizer parameters.

A Simpler voice plays a user-defined region of a sample, which is in turn processed by envelope, filter, LFO, volume and pitch components. Warped samples will play back at the tempo of your Set, regardless of which note you play. Warping in Simpler works in much the same way as it does in audio clips, and bringing a warped clip into Simpler from an audio track, the browser, or your desktop preserves your manual warp settings. When using this expanded view, the parameters in the Controls tab fill Simpler in the Device View.

The Sample Tab displays the sample waveform. Samples can be dragged into Simpler either directly from the browser, or from the Session or Arrangement View in the form of clips.

Samples can be replaced by dragging in a new sample, or by activating the Hot-Swap button in the lower-right corner of the waveform display. This switch is found on the left side of the Sample tab or along the bottom of the expanded sample view.

In Classic Playback Mode, the various sample position controls change which region of the sample you play back. The left flag sets the absolute position in the sample from which playback can start, while the End control sets where playback can end. Start and Length are then represented in percentages of the total sample length enabled by the flags. The Loop slider determines how much of the available sample will loop. This parameter is only active if the Loop switch is enabled.

While this might be exactly the effect you want, it can cause very high CPU loads, particularly when working with the Complex or Complex Pro warp modes. Zooming works the same in all three playback modes. Note: Snapping is based on the left channel of stereo samples. It is therefore still possible, even with Snap activated, to encounter glitches with stereo samples.

The transition from loop end to loop start can be smoothed with the Fade control, which crossfades the two points.

This method is especially useful when working with long, textural samples. The Gain slider allows you to boost or cut the level of the sample.

This parameter is available in all three playback modes. The Voices parameter sets the maximum number of voices that Simpler can play simultaneously.

For example, if your Voices parameter is set to 8, and ten voices are all vying to be played, the two oldest voices will be dropped. Simpler does try to make voice stealing as subtle as possible. With Retrig enabled, a note that is already sustaining will be cut off if the same note is played again.

If Retrig is disabled, multiple copies of the same note can overlap. Note that Retrig only has an audible effect if the sample has a long release time and the number of Voices is set to more than one. The various warp parameters are the same in all three playback modes and are discussed below see In One-Shot Playback Mode, the left and right flags set the available playback region, as they do in Classic Mode, but there are no Loop or Length controls. There is also no Voices control; One-Shot Mode is strictly monophonic.

With Trigger enabled, the sample will continue playing even after the note is released; the amount of time you hold the pad has no effect when Trigger is on. You can shape the volume of the sample using the Fade In and Fade Out controls. Fade In determines how long it takes the sample to reach its maximum volume after a note is played, while Fade Out begins a fade out the specified amount of time before the end of the sample region.

With Gate enabled, the sample will begin fading out as soon as you release the note. The Fade Out time determines how long it will take to fade to silence after release. Snap works similarly to its function in Classic Mode, but only affects the start and end flags because there are no loop options.

The Playback chooser determines how many slices can be triggered simultaneously. Mono is monophonic; only one slice can be played at a time.

 
 

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