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Logic pro x instruments pdf free
All signals below this frequency are allowed to pass. Signals above the frequency are cut. The Low Cut and High Cut parameters work in conjunction with each other to act as a bandpass filter, where signals that fall within the two cutoff ranges are allowed to pass.
Alchemy spectral element effects provide a number of creative options in the spectral synthesis engine. Two effects units are available in the lower half of the spectral parameters shown in the source subpage.
See Logic Pro Alchemy spectral element controls for information on other spectral element parameters. See spectral effect descriptions below. This parameter is common to all spectral effect types.
The parameter name and function vary with each effect type. Note that this effect requires a small amount of calculation time to collect and release a group of frequencies.
As a result, there may be a gap between playing a note and hearing the effect. Tip: Try single note samples with a strong initial attack, such as a piano, and set Mix to a value that introduces the effect as part of the tail of the sound.
When centered 0 , the original frequency balance is used. Blur Blur produces a frequency blurring effect. Tip: Try a melodic loop with pitch variations to best hear the impact of this effect. For example, when used on a loop, higher settings produce a simplified sound with more frequent gaps in the effect output. Cloud Cloud produces what might best be described as a cloud of frequency grains, resulting in a textured chorus effect. Depending on your settings and source material, this can either produce a choppy sound or a smoother one.
Tip: Drum loops are an ideal starting point when learning uses for this effect. This reduces detail and enhances prominent frequencies in the source. Glide Glide creates adjustable, repeating upward filter sweeps that are based on the source content. Note that this effect requires a small amount of calculation time before the results of your adjustments are heard.
Tip: Sources with wide frequency ranges produce a more pronounced filter sweep sound, whereas sources with limited frequencies can result in unique melodic drones as narrow filters sweep across their ranges. Freq Shift Freq uency Shift moves the spectral bins higher or lower in the spectrum, reducing the level of or entirely dropping some frequencies while emphasizing others.
This is a powerful sound design tool that can dramatically alter the sound and can introduce inharmonic overtones. Tip: Try single note samples with a strong initial attack, such as a piano, and blend the mix level so that the effect comes in as part of the tail of the sound.
Start with small adjustments because this parameter has a wide range. Higher frequencies are attenuated. Tip: This effect is highly dependent on the available frequency range in the imported sample. For example, the Alpha and Beta knobs are useful across the entire range with drum loops, whereas the most useful Alpha and Beta ranges are small positive or negative deviations from the center position when used with spoken vocals.
A setting of 1 centered is closest to the source sound. Metallize Metallize produces classic comb filter style effects.
Tip: Experiment with drum loops to clearly hear the impact of the controls. Higher settings emphasize harmonics, creating metallic resonances. Shimmer Shimmer sweeps the frequencies to varying degrees and speeds, imparting either slow frequency shifts or fast shimmering sounds. Tip: Try pure organ samples to clearly see the results of the effect in the real-time spectrogram display, particularly at moderate rate settings.
Tip: Import a bell sample, and start with very low settings to see and hear the impact of controls on the sound. Note that bins are numbered and selected sequentially. Smear Smear averages between blocks of frequencies to create a smoother, more consistent sound. It delivers different results to the Blur effect. Tip: Try melodic loops that have pitch variations to showcase this effect.
Higher settings have less sonic variation, so make small changes. This natural variation in the sound provides a more organic cloud-like effect. Disable to lock the phases of the source, resulting in a tight, metallic sound.
Logic Pro Alchemy pitch correction controls Source components are shown only in advanced view. The parameters in this section are shown when the Pitch button is active in an additive or spectral source subpage.
Higher values result in stronger correction. This is shown as a percentage. Logic Pro Alchemy formant filter controls Source components are shown only in advanced view. The parameters in this section are shown when the Formant button is active in an additive or spectral source subpage. When audio is imported into the additive or spectral engines with the Formant option enabled, the signal is analyzed and resonances in the original signal are extracted and converted into a formant filter shape.
The formant filter scales the amplitude of additive partials or spectral bins over time to recreate the characteristic resonances of the instrument, rather than processing the audio signal like a conventional filter.
This more detailed analysis attempts to determine the resonant frequencies of the source audio data. Higher values can make sounds seem brighter or thinner. Lower values can create a darker, thicker character. Set to lower values to reduce key tracking which may make some sounds playable over a wider keyboard range. The Size knob works in conjunction with the Center parameter. Resonances below the center frequency are shifted upward as the Size knob value is increased.
A corresponding downward shift occurs to resonances above the center frequency. High values smooth and slow down formant changes. Low values exaggerate and speed up changes. Formant filter synthesized parameters The synthesized controls work with any additive or spectral material and do not require the formants to be analyzed on import.
Use these parameters to impose new resonant characteristics on the original signal. Size works in conjunction with the Center knob. Resonances below the center frequency are shifted upward as the Size value is increased. The displayed value indicates position. Whole numbers indicate a particular filter unit, and fractional values indicate a position between filters.
Assign this type to use one of the four filter units as a bypass. Adjust the Select knob to quickly disable synthesized formant processing. The Size knob can be used to stretch the pattern of cuts and boosts up or down the frequency spectrum, or both, depending on the setting of the Center knob.
The negative filter name is used because it recreates the effect of a phase- inverted delayed signal that boosts only odd harmonics, resulting in a hollow sound. This filter has a brighter sound than the negative comb filter. Experiment with each comb to determine the best choice for your sound.
The parallel filters are multipole designs. Signals above or below the set center frequency are attenuated. The Shift knob sets the cutoff frequency. The Size knob changes the filter slope. The frequency band can be moved up or down the frequency spectrum with the Shift knob. The Size knob sets the width of the band notch. The Size knob sets the width of the band. Classic vowel sounds are warmer, and are similar to synthesizer vowel sound filtering.
Smooth variants are more natural-sounding vowel shapes with a gentler filter slope. Each Bright, Classic, and Smooth vowel filter is more of a unique variation on that general sound, with not only brightness differences, but also overall character differences.
Additionally, any vowel filter can be independently modulated, alone or in conjunction with Select knob morphing between filters even from mismatched sets. Use these facilities to dramatically expand your filtering options. Each variation of this complex filter shape has prominent peaks at different frequencies. It is, generally speaking, an open-sounding filter.
This filter shape has gentler midrange and upper midrange peaks with a dominant low-mid resonance. The result is a rounder sound with less brightness and presence than the vowel types above.
This filter shape has gentler midrange and upper midrange peaks with a prominent low-mid resonance. Modify formants in a resynthesized additive guitar sound 1. Select source A, then click the source select field and choose Import Audio from the pop-up menu. Navigate to the Guitars subfolder in the Factory samples folder, and choose a single guitar sample.
When loading is complete, click the Formant button to the right side of the source A window. Note that the upper Analyzed section is turned on. Adjust the Shift knob to move resonances up or down in frequency and to change the timbre. Small amounts of Shift variation work well for subtle changes: try a few semitones in either direction.
Play some very low notes, then some very high notes. Gradually turn down the KTrack knob to reduce key tracking for the formant filter, and note the difference when you replay the high and low notes. Adjust the Size knob value to change the apparent size of the guitar body. Also adjust the Center knob value, and note the effect it has on the tone of the resulting larger or smaller guitar body. Modify formants in a resynthesized spectral drum loop 1. Navigate to the Loops subfolder in the Factory samples folder, and choose a drum loop.
When loading is complete, click the Formant button to the right of the source A window. Adjust the Size knob value to make the drums seem bigger or smaller. Adjust the Smooth knob value to alter the rate of change for the formant filter. Higher values smear the timbre of one drum into the next. Lower values exaggerate changes and create an unusual distortion near the bottom of the knob range.
Create a talking additive sound with synthesized vowel formants 1. Select source A, then turn off the oscillator in the VA section to the right. Click the Additive button, and turn on the additive section. You will hear an additive sawtooth sound if you play some notes.
As an option, increase the Num Partials value. This helps to prevent the sound becoming dull if played in lower registers. Click the Formant button, and turn on the lower Synthesized section. Increase the Select knob value, and play a few notes.
Adjust the Shift knob, the Size knob, and the Center knob, to explore the different timbres available. Switch the order of vowels in the four pop-up menus, and also load different filter types such as Comb.
The parameters in this section are shown when the Granular button is active in a source subpage. The Granular section is available only when you import an audio sample using either granular or sampler mode. Note: The sampler and granular engines are mutually exclusive: you can use one or the other within a single source, but not both together.
You can, however, enable further sources if both engines are required simultaneously. Granular synthesis represents continuous sound as a stream of grains, or tiny pieces of sound. Alchemy generates grains by extracting 2- to millisecond pieces from an audio file. The amplitude of each grain is shaped, along with any pitch and pan modifications, before the grain is sent to the output stream.
Grains can be reordered, time stretched, and pitch shifted. This provides an inexhaustible supply of potential raw material to use as the basis of your sounds. Granular element parameters In addition to the following controls, granular playback is affected by loop modes and by the settings and modulations of the Position and Speed knobs in each source subpage.
Modulations of the granular element update with each new grain. For an example of the impact this has, modulating the source Coarse Tune parameter with an LFO causes the stream of grains to rise and fall in pitch, but does not create pitch sweeps within each grain. If a large Size value is used in conjunction with a low Density value, modulations of source parameters such as pitch may sound stepped, rather than smooth.
The Size and Density parameters interact with each other. When the Density value is 1, a single grain is sent to the output stream. As soon as one grain finishes, the next one is sent. A Size value of msec sends a new grain every msec. Increasing Density to 2 adds a second grain that is sent in between those of the first, resulting in a new grain every 50 msec, assuming a Size value of msec.
The first and second grains overlap each other. Higher Density values inject additional new grains into the output stream. These new grains occur more frequently and overlap more heavily.
Setting Size to around msec and Density to around 5 grains is often suitable for smooth pad sounds with no sharp transients. Setting Size between 40 and 80 msec and Density to around 2 grains is useful for drums and other sounds featuring sharp transients. Small Size values tend to produce a buzz that masks the original pitch of the sample. Large Size values tend to break up the sound.
You can counteract both tendencies by increasing the Density. Note: Also important to the Size and Density parameters is the shape chosen in the Grain Shape pop-up menu. This can have a significant or subtle impact on sonic artifacts that may be introduced in the stream of grains. The source Stereo button must be on for RPan to have an effect. Taps retrigger the attack phase of the source. Note: Taps that fall within a looped area are retriggered on each loop cycle.
Values are shown as a percentage of the overall sound duration. Set to zero to trigger taps in quick succession at the sound end point. The source Stereo button must be on for Stereo Offset to have an effect. At a basic level, this applies a small fade-in and fade-out to each grain, but some shapes may have a more significant impact, depending on the current Size and Density values and the source material. You can also step through the available grain shapes with the Previous and Next buttons the arrows.
This function is primarily intended to reduce or remove glitches, clicks, and crackles in the playback of a stream of grains, but it can introduce buzzy gaps between grains and can affect the tonality of grains. There are no fixed rules when it comes to the choice of grain shape, given the infinite variety of source audio material. Therefore, you may want to experiment to achieve the required results.
The parameters in this section are shown when the Sampler button is active in a source subpage. The sampler section is available only when you import an audio sample using either granular or sampler mode.
The sampler section allows audio files, known as samples, to be played directly. Samples played at a higher pitch than the original play back at a faster speed. Samples played at a lower pitch than the original play back at a slower speed. The sample waveform is displayed in the center. A progress bar indicates the current playback position for the most recently triggered note. When multiple elements are used in a source, use this control to set the relative level of the sampled component.
The parameters in this section are shown when the VA Virtual Analog button is active in a source subpage.
When you click the Name bar File button, and choose Initialize Preset from the pop-up menu to initialize Alchemy to default settings, the VA element is automatically enabled. Basic saw, sine, square, and triangle and many specialized waveforms are provided. You can also step through the available waveforms with the Previous and Next buttons the arrows. When multiple elements are used in a source, use this control to set the relative level of the oscillator component.
When a square wave is active, Symmetry acts as a pulsewidth control. These have different spectral characteristics that can be further refined with filters.
You can step through the available waveforms with the Previous and Next buttons the arrows. When multiple elements are used in a source, use this control to set the relative level of the noise component.
All frequencies above this value are allowed to pass. All frequencies below are attenuated. All frequencies below this value are allowed to pass.
All frequencies above are attenuated. The Low Cut and High Cut parameters work in conjunction with each other to act as a bandpass filter, where the noise signal that falls within the two cutoff ranges is allowed to pass. Logic Pro Alchemy source modulations Source components are shown only in advanced view. Parameters that have a modulation assignment are indicated by an orange arc around the control. Note: Parameters that are morphed and have a modulation assignment show both an orange and green arc around the control.
This section focuses on Position, which is a modulation target. The principles discussed apply equally to other source parameter targets. Position determines the playback position of audio data. When modulated, the playback path through the audio data is controlled by the selected modulation source. In sampler mode, the note-on modulation value determines the initial offset for the play position within the audio data. Beginning at that position, the rest of the sound plays in a normal manner, although looped as if the Loop mode is set to All.
In additive, spectral, or granular mode, Position can be continuously modulated forward or backward at any rate including zero. Create tempo-synced loops by modulating position Synchronized playback of looped audio with the Logic Pro X tempo is easy to achieve by modulating the Position parameter. This technique is possible with any synthesis method that permits continuous modulation of Position. This example uses the granular engine, but the same technique can be applied to the additive and spectral engines.
When Position is modulated and Speed has a value greater than zero, the playback path is determined by a combination of modulation value whenever this value changes and the normal path at a rate determined by Speed whenever the modulation value is static.
In source A, import a rhythmic or melodic sample that loops evenly. You will hear that playback is frozen at the very beginning of the sample. Do one of the following:. Note the orange arc that appears around the Position knob. This indicates that the parameter has a modulation assignment. Also in the LFO, turn off the Bipolar button. This routing increases Position smoothly so that the entire sample plays back from beginning to end, then jumps immediately back to the beginning and continues to loop.
Finally, adjust the Logic Pro X tempo as you play additional notes to confirm that the loop is properly synchronized. Logic Pro Alchemy morph controls Source components are shown only in advanced view. Click the Advanced button to switch to advanced view, then click the Morph button to view and use the morph controls. The morph controls determine how the four Alchemy sources interact. There are two basic types of interaction. This is equivalent to turning the Amp knobs in each source to attain the desired mix.
If you crossfade from a source with a high Coarse Tune setting to a source with a low Coarse Tune setting, the high source fades out as the low source fades in. In the middle of the crossfade you hear both sources. If you morph from a source with a high Coarse Tune setting to a source with a low Coarse Tune setting, you hear a single sound during the morph.
The sound tuning falls smoothly from the high value to the low one. Morphing provides more scope than simple crossfades between sources. It also allows cross-synthesis, where you can combine different aspects of different sound elements. For example, you could apply the formants or other characteristics of an additive source to the spectral element of another source. See the tutorials found in Logic Pro Alchemy elemental morphs overview. Morphed parameters are indicated by a green arc around the control.
Parameters that are morphed and have a modulation assignment show both an orange and green arc around the control. Parameter settings are shared across all morphed sources, which means that changing a parameter in one source results in the corresponding parameter being changed for all morphed sources.
Note: Parameters that do not directly participate in the morph, including most buttons and pop-up menus, are indicated with a lock icon displayed at the top left of the control the lock icons are shown only in source subpages.
Where there is a parameter pairing of an On button and a pop-up menu, only the button shows the lock icon. Neither parameter participates in the morph. Regions of each source encompassed by corresponding warp markers are time- aligned in the morph. See Logic Pro Alchemy zone waveform editor.
Also controls VA morph position if the VA element is active. Also controls sampler morph position if the sampler element is active.
Also controls morphing of the source filter knobs. These controls morph the timing of the sound. In cases where source A has a short attack and source B has a long attack, for example, the length of the attack varies as you change the X knob. Turn off for the best morphing quality.
As an example, Auto Align corrects the timing of words of four spoken voice samples saying the same phrase in each of the four morphed sources. Auto Align is automatically turned off when you set warp markers manually. Regions of each source encompassed by corresponding warp markers are time-aligned in the morph. See Logic Pro Alchemy elemental morphs overview. Use Elements to view and edit the X values of five parameters.
Morphs affect only the chosen group of sources. This also controls the sampler morphing position if the sampler element is active. Turn off for higher morphing quality. For example, Auto Align corrects the timing of words of four spoken voice samples saying the same phrase in each of the four morphed sources. Drag the point to change the X or Y value, or both. Drag each point to change the corresponding X and Y values. Edit buttons are shown on all source subpages.
Shown at the top right of all edit windows. By default, the Main edit window is shown. Further Additive and Spectral edit windows can be opened by clicking the buttons at the top of the window. These windows provide additional parameters that let you precisely edit and sculpt your sounds. The Main edit window is divided into three areas that interact with each other.
You can edit parameters graphically in the keymap or waveform editor or can use corresponding fields and other parameters in the inspector. This area interacts with the zone parameters in the inspector. See Logic Pro Alchemy keymap editor. The source edit window is opened by clicking the Edit button on any source subpage.
Click the close window icon X to close the source edit window. The source inspector is divided into three main parameter groupings: global and source parameters, group parameters, and zone parameters.
Click the X icon at the top right of the active window to close it. You can also click the Previous and Next buttons the arrows to step through available waveform data. All other sources are muted. Logic Pro Alchemy inspector group controls Source components are shown only in advanced view. Click the close window icon X at the top right to close the window.
The new group name is added below existing group names in the Group list shown under the Group pop-up menu. A sequentially assigned number is appended to the new group name. Control-click a group name to choose the Delete command. Attack triggers group zones when note-on messages are received. Release triggers group zones for note-off messages.
This mode is useful for instruments with a distinctive end-of-note sound such as a key click or hammer thump. Such sounds can be included as a separate group with release triggering enabled.
You can also use this feature creatively to add abstract reverb tails or to create a pad sound that changes dramatically during the release stage. Apply a fade in time value ranging from 0 to for note-off events when the release trigger mode is active. This parameter is primarily intended for use in conjunction with release triggering, to create a crossfade between the main body of the note and the release sample.
A common use of this feature is to create a group containing an open and a closed hi-hat sample. If you set group polyphony to one, either the closed or open hi-hat sample can play, but not both at the same time. You can create and combine multiple rules using boolean logic. The normal state for any newly created group is a single rule set to Always, unless that group was defined as a Round Robin group in the Import browser Dropzone.
Click the field to choose a rule. This adds a new rule pop-up menu below the first and displays a Logic pop-up menu to the right. Choose Delete from the pop-up menu to remove a rule. At least two round robin groups must exist for this parameter to have an effect. Each group is assigned a different value. If two groups are assigned to the same value, both zones are triggered simultaneously.
Each note-on sequentially triggers a round robin group from lowest to highest. Once the highest group number is triggered, the sequence starts again from the lowest group number. Drag vertically in the field or use the arrows to set a value. No Order pop-up menu is shown. You can also assign keyswitching to a MIDI note or range of notes which prevents the group from triggering until one of these notes is played.
Two pop-up menus are displayed, where you can choose keyswitch conditions. For example, set the first field to Snap2 and the second field to Snap4, which results in the group being triggered only when the Transform pad is at position 2, 3, or 4. Assign multiple groups to different ranges to switch between up to eight different groups.
See Logic Pro Alchemy Transform pad. This knob is visible only when the Keysw options are chosen in these menus. For example, set the first field to Keysw1 and the second to Keysw5, which results in the group being triggered when the Keyswitch knob is set to a value that falls within this range. Up to 10 Keyswitch knob positions are available, enabling you to switch between groups with the knob.
Because the Keyswitch knob is available as a modulation target, this lets you create complex automated group switches. Play a note in this range to switch to a group, which remains active until another group is chosen. Three pop-up menus are displayed where you can choose the controller type and set controller values. You can set other groups with the same type but with a different control range to switch between groups with a single controller.
Because the Control 1 knob used in the example is available as a modulation target, this lets you create complex automated group switches. Drag vertically in the fields or use the arrows to set a value. The different logic conditions result in different outcomes.
Crossfade to a different group of samples on note-off 1. Select a sample or multiple samples representing the main sustain portion of your sound, and import using any of the available import modes. Alchemy analyzes each sample to determine the root pitch if not defined in the filename , set the root key, key range, and velocity range for each sample zone such that they span the entire keyboard and the entire dynamic range, and add all zones to a group named Group 1.
Select a sample or multiple samples representing the release portion of your sound, and import the import mode is automatically set to match the existing group. Alchemy again analyzes each sample and adds all zones to a group named Group 2. Double-click Group 2, then click the Trig field and change it to Release. Zones in Group 2 will now trigger when you release each key, playing over zones in Group 1 which continue to sound until AHDSR1 reaches the end of the release stage.
Double-click Group 1 in the list, then click the Fade field and change it to a value other than 0. Zones in Group 1 will now fade out when the note is released, allowing Group 2 to be heard during the release stage of the sound. Higher Fade values result in slower fades.
As an option, double-click Group 2 in the list, then click the Fade field and change it to a value other than 0. Zones in Group 2 will now fade in when the note is released, creating a crossfade between Groups 1 and 2 at note-off.
Higher values result in slower fades. Fading in the release group may be unnecessary if your release samples already have a natural fade in at the start, however, or undesirable if a percussive transient is required at note-off. Try to set Fade values for Group 1 and 2 to an identical small value to create a sudden but click-free crossfade at the end of each note. Create random round robin variations 1. Select a sample or multiple samples representing the second of your round robin variations, and import the import mode is automatically set to match the existing group.
Alchemy will again analyze each sample and add all zones to a group named Group 2. Any notes you play will now randomly trigger either group 1 or group 2, but not both together. Note that if you play a chord, each individual note is randomly assigned to one of those groups. Repeat steps 7 to 9 as needed to configure a group for each further variation you require.
Assign the source Keysw knob to switch between groups 1. Click the Rule field, and change it from Always to Keyswitch. Click the first range field, and change it to Keysw1. The second range field also changes to the same value. As an option, you can click the second range field and increase the value to specify a range of values that will trigger this group, instead of just a single value.
Select a sample or multiple samples representing the second of your variations, and import the import mode is automatically set to match the existing group. Click the Rule field below Group 2, and change it from Always to Keyswitch. Click the first range field, and change it to the first unused Keysw value.
If the second range field for Group 1 is set to Keysw3, choose Keysw4. As an option, you can click the second range field and increase the value to specify a range of values that will trigger Group 2, instead of just a single value.
Click the X symbol at the top right to close the source edit window. A new Keysw knob is visible in the source pane, to the left of the Keyscale field. Rotate the Keysw knob to switch between the groups you created. Control-click this knob to add modulation routings from the shortcut menu. Create round robin variations for just one Transform pad position 1. It is automatically assigned the number 1 in the sequence, and a second rule set to Always is added below.
Leave the first range field for rule 2 set to Snap1, and change the second range field to Snap7. Aritz Zabaleta. Roberto Palazzolo. Log in with Facebook Log in with Google. Remember me on this computer. Enter the email address you signed up with and we'll email you a reset link. Need an account? Click here to sign up. Download Free PDF. Logic Pro X Instruments. Related Papers. Musical Mindstorms. Evolutionary computation applied to the control of sound synthesis.
Automatic sound synthesizer programming: techniques and applications. Audio Plug-Ins Guide. You then choose the desired recording settings and adjust the recording level of your sound source to avoid distortion. In the following exercises, you will set up Logic to prepare for a music recording. The microphone transforms sound pressure waves into an analog electrical signal. The microphone preamp amplifies the analog electrical signal. A gain knob lets you set a proper recording level and avoid distortion.
The audio interface sends the digital data stream from the converter to the computer. Logic Pro saves the incoming data as an audio file displayed on the screen by a waveform representing the sound pressure waves. To convert the analog signal into a digital data stream, the digital converters sample the analog signal at a very fast time interval, or sample rate. The sample rate identifies how many times per second the audio is digitally sampled.
The bit depth identifies the number of data bits used to encode the value of each sample. The sample rate and bit depth settings determine the quality of a digital audio recording.
Logic does not exert any influence over the quality of your recordings. Also, most modern Mac computers include a built-in audio interface. Many Mac notebook computers and iMac computers even have internal microphones.
Although those microphones are generally not intended to produce professional-quality recording, you can use the internal microphones to perform the exercises in this lesson in the absence of an external microphone.
By default, Logic records with a bit depth of 24 bits, which is fine for most uses. However, you may need to use different sample rates for different projects. Playing an audio file at the wrong sample rate will result in the wrong pitch and tempo, much like playing an audiotape or vinyl record at the wrong transport speed. The Project Settings window opens, and you can see your Audio settings. By default, the sample rate is set to To determine which sample rate to choose, consider the sample rate of any prerecorded material you will use such as samples and the sample rate of the target delivery medium.
Some producers who make intensive use of Traditionally, music is recorded at Choosing an Audio Interface In most situations, Logic automatically detects an audio interface when you connect it to your Mac and asks if you want to use that interface. If you choose to use it, Logic selects that interface as both an input and output device in its audio preferences.
The Audio preferences appear. The Output Device is the device connected to your monitors or headphones. The Input Device is the device into which you plug your microphones or instruments. If you do not have an audio interface connected to your Mac, choose from the built-in output and input devices. If you choose a new output or input device, Logic automatically reinitializes the Core Audio engine when you close the window. Recording a Single Track In this example, you will record a single instrument.
The exercise describes recording an electric guitar plugged directly into an instrument input on your audio interface, but feel free to record your voice or any instrument you have.
Preparing a Track for Recording To record audio, you first have to create a new audio track, select the correct input the input number on your audio interface where the guitar is plugged in , and enable that new track for recording.
When adding tracks, the new tracks are inserted below the selected track. To create a new track at the bottom of the Tracks area, you first need to select the bottom track. The New Tracks dialog appears. You can record-enable the track by selecting the Record Enable option below the Output menu; however, in some situations creating a recordenabled track may produce feedback.
You will later take precautions to avoid feedback and then record-enable the track from the track header. A new audio track set to Input 1 is created. Logic automatically assigns the new track to the next available channel. Since six audio tracks were created when you dragged Apple Loops in Lesson 1, the new track is assigned to the Audio 7 channel and is automatically named Audio 7.
More descriptive names will help you identify files in the future. The new track has a generic audio waveform icon. You can now hear your guitar and see its input level on the Guitar channel strip meter in the inspector. This delay is called latency. You can monitor the audio routed to record-enabled tracks while Logic is stopped, playing, or recording.
Otherwise, you will be monitoring the signal twice, resulting in a flangy or robotic sound. To emulate the character a guitar amp can give to a guitar sound, you can use Amp Designer, a guitar amplifier modeling plug-in. Note that you are still recording a dry guitar sound. The effect plug-in processes the dry audio signal in real time during the recording and playback.
Recording a dry signal means that you can continue fine-tuning the effect plug-ins or exchange them for other plug-ins after the recording is completed. Amp Designer opens. Here, you can dial in a sound or choose a preset. You can now hear your guitar processed through Amp Designer. Adjusting the Recording Level Before recording, make sure you can monitor the sound through Logic, and then adjust the source audio level to avoid overloading the converters.
On the channel strip, look at the peak level meter, and make sure it always stays below 0 dBFS decibels full scale, the unit used to measure levels in digital audio ; a level above 0 dBFS would indicate that you are clipping the input of your converter. Keep in mind that you need to adjust the audio level before the converter input by using your microphone preamp gain knob.
Allow some headroom, especially if you know that the artist might play or sing louder during the actual recording. Working with a low-level recording is better than clipping the input. Some interfaces also support other input settings, such as phantom power, hi-pass filter, and phase.
If the Gain knob is dimmed, it means that the feature is not supported by your audio interface. Make sure the peak sits comfortably below 0 dBFS: the wider the dynamic range of the source, the more headroom it needs to avoid clipping.
When your signal peaks below —2. When it peaks between —2. When it peaks above 0 dBFS, the peak level meter turns red to indicate the audio is clipping. Tuning the Instrument Making sure an instrument is in tune before recording is always a good idea. The Tuner opens. Checking the Balance Now that the guitar is tuned, you can practice the performance and make sure that you can hear yourself and the other instruments comfortably.
If the guitar is now too loud or too soft in comparison to the other tracks, in the inspector, drag the volume fader on the Guitar channel strip to adjust the monitoring level, or drag the volume slider in the Guitar track header.
Recording Audio You have set the desired sample rate, adjusted the recording and monitoring levels, inserted a plug-in to emulate the sound of a guitar amp, and tuned the instrument. You are now ready to start recording.
The playhead is positioned at bar If you need to adjust the position of the playhead, drag it left or right. The playhead and the LCD display in the control bar both turn red to indicate that Logic is recording. The playhead jumps one bar earlier and gives you a four-beat count-in with an audible metronome click before the recording starts. You will learn how to alter both the metronome and the count-in settings later in this lesson.
The new recording, Guitar 01, appears as a blue-shaded audio region. To the name of the track, Logic appends the number of the recording. The playhead jumps to the beginning of the selected region and playback starts. If you are not happy with your new recording, you can delete it and start over. In the Finder, the audio file is moved from inside the project package to the Trash.
The audio file stays in the Project Audio Browser and is still present inside the project package, allowing you to later drag it back to the workspace if necessary. This alert appears only when you try to delete a recording made since you most recently opened the project. When deleting an audio region that was previously recorded, the behavior corresponding to the Keep option is automatically applied and an alert does not appear. You will keep your recording so you can experiment with recording additional takes in the next exercise.
Recording Additional Takes When recording a live performance, musicians can make mistakes. Rather than deleting the previous recording and repeatedly recording until you get a flawless performance, you can record several takes repeat performances of the same musical part and later choose the best take, or even combine the best parts of each take to create a comp composite take.
To preserve multiple takes in Logic, you can record new performances over previous ones. The new recording in red appears to be recorded over the previous blue audio region.
Both the original recording Take 1 and the new recording Take 2 have been saved into a take folder. The take folder is on the Guitar track. It is currently open, so the two takes you recorded are displayed on subtracks below. By default, the take folder plays the most recent take you recorded: Take 2, in this case. The previous take, Take 1, is dimmed and muted. The track is disarmed, and you can no longer hear the sound coming from Input 1 on your audio interface.
The take folder now contains three takes. It plays back the most recent one, Take 3, while the two previous ones, Take 1 and Take 2, are muted. Recording in Cycle mode allows you to repeatedly record a single section, thereby creating a new take for each pass of the cycle. When you stop recording, all the takes are saved inside a take folder.
The Guitar track is automatically record-enabled. The playhead jumps a bar ahead of the cycle for a one-measure count-in, and starts recording the first take.
When it reaches bar 9, the end of the cycle area, it jumps back to bar 5 and starts recording a new take. Logic keeps looping the cycle area, recording new takes until you stop recording. Record two or three takes. All the takes recorded in Cycle mode are packed into a take folder. The Guitar track is automatically disabled for recording.
To keep the last take of a cycle recording, make sure you stop the recording more than one bar after the beginning of the cycle area. The take folder closes. Doing so allows you to record several instruments at once, placing each instrument on a separate track, so that you can later adjust their volumes and stereo positions or process them individually.
You first create the desired number of tracks, making sure that each track is assigned to a different input number that corresponds to the input number on your audio interface where the microphone is plugged in.
In the following exercise, you will record two mono tracks at the same time, which you can do using the built-in Mac audio interface. To record more than two tracks at once, you need an audio interface with more than two inputs. The exercise describes recording an acoustic guitar on Input 1 and a vocal microphone on Input 2.
When creating multiple tracks, selecting Ascending automatically sets the inputs or outputs to ascending settings. In this case, you will create two tracks, so the first will be assigned to Input 1 and the second to Input 2. Make sure that you took precautions to avoid feedback, as explained at the beginning of this lesson; this time you will create record-enabled tracks. Two new tracks are added at the bottom of the Tracks area and automatically assigned to the next available audio channels Audio 8 and Audio 9.
Their inputs are set to Input 1 and Input 2, and both are record-enabled. The multitrack recording starts, and after a one-measure count-in, you see the red playhead appear to the left of the workspace, creating two red regions, one on each record-enabled track. You now have a new blue-shaded audio region on each track. You can use the same procedure to simultaneously record as many tracks as needed.
If the tracks already exist in the Tracks area, make sure you assign them the correct inputs, record-enable them, and start recording. Punching In and Out When you want to correct a specific section of a recording—usually to fix a performance mistake—you can restart playback before the mistake, punch in to engage recording just before the section you wish to fix, and then punch out to stop recording immediately after the section while playback continues.
This technique allows you to fix smaller mistakes in a recording while still listening to the continuity of the performance. At any time, you can open the take folder and select the original recording. There are two punching methods: on the fly and automatic.
Punching on the fly allows you to press a key to punch in and out while Logic plays, whereas automatic punching requires you to identify the autopunch area in the ruler before recording. Punching on the fly is fast but usually requires an engineer to perform the punch-in and punch-out while the musician is performing.
Automatic punching is ideal for the musician-producer who is working alone. Assigning Key Commands To punch on the fly, you will use the Record Toggle command, which is unassigned by default. Click the disclosure triangle next to Global Commands. The Key Commands window lists all available Logic commands and their keyboard shortcuts, if any.
When looking for a specific functionality in Logic Pro X, open the Key Commands window and try to locate the function using the search field. A command likely exists for that functionality that may or may not be assigned. When Learn by Key Label is selected, you can press a key, or a key plus a combination of modifiers Command, Control, Shift, Option , to create a keyboard command for the selected function. An alert indicates that the R key is already assigned to the Record command.
You could click Replace to assign R to Record Toggle, but then Record would no longer be assigned to a keyboard shortcut. Control-J is now listed in the Key column next to Record Toggle, indicating that the command was successfully assigned. Punching on the Fly You will now use the Record Toggle key command you assigned in the previous exercise to punch on the Vocals track the bottom track in your Tracks area. When punching on the fly, you may first want to play the performance to determine which section needs to be re-recorded, and to be ready to punch in and out at the desired locations.
Position your fingers on the keyboard to be ready to press your Record Toggle key command when you reach the point where you want to punch in. The playhead continues moving, but Logic is now recording a new take on top of the previous recording. Keep your fingers in position to be ready to punch out. The recording stops while the playhead continues playing the project. On the Vocals track, a take folder was created.
It contains your original recording Take 1 and the new take Take 2. A comp is automatically created Comp A that combines the original recording up to the punch-in point, the new take between the punch-in and punch-out points, and the original recording after the punch-out point. Fades are automatically applied at the punch-in and punch-out points.
You will learn more about fades in Lesson 3. The take folder disappears, and you once again see the Vocals 01 region on the Vocals track. Punching on the fly is a great technique that allows the musician to focus on his performance while the engineer takes care of punching in and out at the right times.
On the other hand, if you worked alone through this exercise and tried to punch in and punch out while playing your instrument or singing, you realize how challenging it can be. When working alone, punching automatically is recommended. Punching Automatically To prepare for automatic punching, you enable the Autopunch mode and set the autopunch area. Setting the punch-in and punch-out points in advance allows you to focus entirely on your performance during recording.
First, you will customize the control bar to add the Autopunch button. The ruler becomes taller to accommodate for the red autopunch area. The autopunch area defines the section to be re-recorded.
You can define the autopunch area with more precision when you can clearly see where the mistakes are on the audio waveform. Logic zooms in, and the selected region fills the workspace. Here we have a vocal recording in which the two words around bar 3 need to be re-recorded. Listen while watching the playhead move over the waveform to determine which part of the waveform corresponds to the words you need to replace.
You can drag the edges of the autopunch area to resize it, or drag the entire area to move it. Red vertical guidelines help you align the punch-in and punch-out points with the waveform. Playback starts. When the playhead reaches the punch-in point the left edge of the autopunch area , the Record button turns solid red and Logic starts recording a new take. When the playhead reaches the punch-out point the right edge of the autopunch area , the recording stops but the playback continues.
A take folder, Vocals: Comp A, is created on the track. Logic zooms out so you can see the entire take folder filling the workspace.
Just as when you punched on the fly in the previous exercise, a comp is automatically created that plays the original recording up to the punch-in point, inserts the new take between the punch-in and punch-out points, and continues with the original recording after the punch-out point.
When a marquee selection is present, starting a recording automatically turns on the Autopunch mode, and the autopunch area matches the marquee selection. Recording Without a Metronome Musicians often use a tempo reference when recording.
In most modern music genres, when live drums are used, drummers record their performance while listening to a metronome or a click track. When electronic drums are used, they are often recorded or programmed first, and then quantized to a grid so that they follow a constant tempo.
The other musicians later record their parts while listening to this drum track. Still, some musicians prefer to play to their own beat and record their instrumental tracks without following a metronome, click track, or drum track. When recording audio in Logic, you can set up Smart Tempo to analyze a recording and automatically create a tempo map that follows the performance so that the notes end up on the correct bars and beats.
Subsequent recording or MIDI programming can then follow that tempo map, ensuring that all tracks play in sync. An empty project template opens, and the New Tracks dialog opens. To make Logic analyze the audio recording and create a corresponding tempo map, you should set the Project Tempo mode to Adapt.
The orange color indicates that those parameters will be affected by a new recording. Get ready to record. Because the Project Tempo mode is set to Adapt, the metronome does not automatically play unlike the Project Tempo mode set to Keep mode. You no longer need it! Try playing something that has an obvious rhythmic quality to it, such as a staccato rhythm part in which you can clearly distinguish the individual chords or notes.
During the recording, Logic displays red vertical lines over the recording when it detects beats. An alert offers to open the File Tempo Editor so you can preview the recording and adjust the positions of the beat markers that Logic created while analyzing the file.
In the Global Tempo track, you can see multiple tempo changes. In that case, perform this exercise again, making sure you can hear a strong rhythmic reference in your recording.
For example, try tapping a very basic beat with your fingers in front of the microphone. You have recorded a rubato performance without listening to a timing reference. Logic automatically detected your tempo changes and applied them to the project tempo.
Some settings do not affect the quality of the audio recording but can alter the behavior of your project during recording or change the audio file format used for recordings. The next few exercises will show you how those settings affect the audio recording process and explain how to modify them. Setting the Count-In The count-in is the time you have to prepare yourself and get in the groove before the actual recording begins.
The take folder is deleted. Until now, every time you pressed Record, the playhead jumped to the beginning of the previous measure so you could have a four-beat count-in.
However, sometimes you may want to start recording without a count-in. The playhead starts from its current position, and Logic starts recording right away.
At other times, you may need a longer count-in, or you may want Logic to count in for a specific number of beats. The audio region is removed from the workspace, but the audio file is still in the project folder.
The playhead jumps two bars ahead to bar 3, and playback starts. When the playhead reaches bar 5, Logic starts recording. Setting the Metronome By default, the metronome is turned off during playback and automatically plays during recording. In this exercise, you will change the default behaviors using the Metronome button and later go into the Metronome settings to adjust its sounds. The metronome is on. The metronome is off. The metronome is back on. You now have inverted the default behavior: the metronome is on during playback and is automatically turned off during recording.
The Metronome Settings window opens. There are settings for two metronomes: Audio Click also known as Klopfgeist, which is German for knocking ghost , which you are using, and MIDI Click, which is now off. Under the name of each metronome, you can adjust the pitch and velocity of the notes playing on each bar and beat. The metronome sounds a little low compared to the drum loop on track 1.
In fact, you can hear it only when no drum hit occurs on that beat. At the bottom of the Metronome Settings window, you can drag a couple of sliders to adjust the sound of the metronome.
The metronome sound changes, and you can start hearing a pitch. When a project already contains a drum track, you may need the metronome only during the count-in to get into the groove before the song starts. You hear the metronome for one measure, and then it stops playing as the song and the recording start at bar 1.
It places a number of samples in an input buffer for recording and in an output buffer for monitoring. When a buffer is full, Logic processes or transmits the entire buffer. The larger the buffers, the less computing power is required from the CPU. The advantage of using larger input and output buffers is that the CPU has more time to calculate other processes, such as instrument and effects plug-ins. The drawback to using a larger buffer is that you may have to wait a bit for the buffer to fill before you can monitor your signal.
That means a longer delay between the original sound and the one you hear through Logic, a delay called roundtrip latency. Usually, you want the shortest possible latency when recording and the most available CPU processing power when mixing so that you can use more plugins. The Audio preferences pane opens. When choosing a different audio device, make sure you click Apply Changes to update the Resulting Latency value displayed.
The latency is now shorter. If your Mac has a multicore CPU, you can see a meter for each core. You can monitor the amount of work each core is doing. When the CPU works harder, you might hear pops and crackles while the song plays. When playing the project becomes too much work for the CPU, playback stops and you will see an error alert. Deleting Unused Audio Files The Project Audio Browser shows all the audio files and audio regions that have been imported or recorded in your project.
During a recording session, the focus is on capturing the best possible performance, and you may want to avoid burdening yourself with the decision making that comes with deleting bad takes. You may also have several unused audio files in the Project Audio Browser that make the project package or folder bigger than it needs to be.
In this next exercise, you will select and delete all unused audio files from your hard drive. The audio data in the audio file stays intact, and the regions merely point to different sections of the audio file.
You will learn more about nondestructive editing in Lesson 3. If a Delete alert appears, select Keep and click OK. The regions are removed from the workspace, but their parent audio files are still present in the Project Audio Browser. All the audio files that do not have an associated region in the workspace are selected. While the region plays, a small white playhead travels through the regions.
Once you feel satisfied that the selected audio files do not contain any useful material, you can delete them. An alert asks you to confirm the deletion. The audio files are removed from the Project Audio Browser. In the Finder, the files are moved to the Trash. You are now ready to tackle many recording situations: you can record a single track or multiple tracks, add new takes in a take folder, and fix mistakes by punching on the fly or automatically.
You know where to adjust the sample rate, and you understand which settings affect the behavior of the software during a recording session. And you can reduce the file size of your projects by deleting unused audio files—which will save disk space, and download and upload time should you wish to collaborate with other Logic users over the Internet. What two fundamental settings affect the quality of a digital audio recording?
In Logic, where can you find the sample rate setting? What precaution must you take before record-enabling multiple tracks simultaneously? In Autopunch mode, how do you set the punch-in and punch-out points? Describe an easy way to access your Metronome settings.
Describe an easy way to access your count-in settings. In the Project Audio Browser, when selecting unused files, what determines whether a file is used or unused? The sample rate and the bit depth 2. Make sure the tracks are assigned different inputs. Adjust the left and right edge of the autopunch area in the middle of the ruler.
Control-click the Metronome button, and choose Metronome settings. The CPU works less hard so you can use more plug-ins, but the roundtrip latency is longer. An audio file is considered unused when no regions present in the workspace refer to that file.
Goals Assign Left-click and Command-click tools Edit audio regions nondestructively in the workspace Add fades and crossfades Create a composite take from multiple takes Import audio files Edit audio regions nondestructively in the Audio Track Editor Align audio using the Flex tool Audio engineers have always looked for new ways to edit recordings.
In the days of magnetic recording, they used razor blades to cut pieces of a recording tape and then connected those pieces with special adhesive tape.
They could create a smooth transition or crossfade between two pieces of magnetic tape by cutting at an angle. Digital audio workstations revolutionized audio editing. The waveform displayed on the screen is a visual representation of the digital audio recordings stored on the hard disk.
The ability to read that waveform and manipulate it using the Logic editing tools is the key to precise and flexible audio editing. In this lesson, you will edit audio regions nondestructively in the workspace and the Audio Track Editor, and add fades and crossfades.
You will open a take folder and use Quick Swipe Comping to create a single composite take. Even as your ability to read waveforms and use the Logic editing tools develops, never forget to use your ears and trust them as the final judge of your work. Assigning Mouse Tools Until now, you have exclusively worked with the default tools. You have also used keyboard modifiers such as Control-Option to choose the Zoom tool, and changed the pointer to tools such as the Resize or Loop tools. When editing audio in the workspace, you will need to access even more tools.
In the Tracks area and in various editors , two menus are available to assign the Left-click tool and the Command-click tool. Previewing and Naming Regions During recording sessions, helping the talent produce the best possible performance often takes priority over secondary tasks such as naming regions.
In this exercise, you will assign tools to the mouse pointer. You will use the Solo tool to preview the audio regions on the new Guitar track, and apply the Text tool to rename them.
You can hear a region play back in solo mode by placing the Solo tool over the region and holding down the mouse button. In the control bar, the Solo button turns on, and the LCD display and the playhead both turn yellow. The region is soloed, and you can play back starting from the location where you placed the Solo tool.
You can also drag the Solo tool to scrub the region. You can change the playback speed or direction by dragging the Solo tool to the right or to the left. You can hear that the guitar is playing single, muted notes, so you will give it a descriptive name based on those notes. If you hold down Command when your pointer is over a region, it changes to the Text tool. A text field appears, in which you can enter a new name for the region.
You can hear some dead notes at the beginning of this take folder, and about a bar of funk rhythm guitar in bar You will edit this take folder later in this lesson. In those regions, the guitar sustains chords, so you will name the regions after the chord names. Instead of moving back and forth from the workspace to the tool menus in the Tracks area menu bar, you can press T to open the Tool menu at the current pointer position.
A Tool menu appears at the pointer position. This key command will save you a lot of trips to the title bar. You can also Command-click a tool in the pop-up Tool menu to assign it to the Command-click tool.
The Tool menu opens and closes, and the Left-click tool reverts to the Pointer tool. Both tools are back to their default assignments: the Pointer tool for the Left-click tool and the Marquee tool for the Command-click tool.
Editing Regions in the Workspace Editing audio regions in the workspace is nondestructive. Regions are merely pointers that identify parts of an audio file. When you cut and resize regions in the workspace, only those pointers are altered. No processing is applied to the original audio files, which remain untouched on your hard disk. As a result, editing in the workspace provides a lot of flexibility and room for experimentation because you can always adjust your edits at a later date.
In this next exercise, you will edit the Muted Single Notes region on the Guitar track. In the Snap menu, a checkmark appears in front of the modes you choose. The help tag shows that the region length is now 4 0 0 0. You will now repeat the simple motif in the last two bars of the Muted Single Notes region a couple more times, from bars 9 to 13, where the synthesizers play.
The Command-click tool is now the Marquee tool, and the Left-click tool is the Pointer tool. This is a very powerful tool combination when editing audio in the workspace.
You can select a section of an audio region with the Marquee tool, and move or copy that selection using the Pointer tool. The section you selected with the Marquee tool is highlighted. The playhead jumps to bar 7 and plays the selection. It corresponds exactly to the two-bar pattern of the guitar you are going to copy. Option-dragging a marquee selection automatically divides, copies, and pastes the selection to a new location regardless of region boundaries.
In this example, the two-bar guitar pattern is copied and pasted at bar 9. Remember to release the mouse button first and the Option key second. When the mouse button is released, the original region is automatically restored. The guitar plays a melodic riff with high notes when it first comes in, and then it plays more discretely throughout the following sections, leaving room for the two synths to shine.
Still, you can bring back a little bit of the excitement just before the breakdown at bar This last region brings back a welcome variation to the monotonous pattern that the guitar has been playing for the past five bars, returning in time to lead to the break in the next section. Now you know how to select the desired material within a region and move or copy that material anywhere on the track.
Comping Takes In the previous lesson, you recorded several takes of a guitar performance and packed them into a take folder. Now you will learn how to preview those individual takes and assemble a composite take by choosing sections from multiple takes, a process called comping. Comping techniques are useful when you have recorded several takes of the same musical phrase, each with its good and bad qualities. In the first take, the musician may have messed up the beginning but played the ending perfectly.
And in the following take, he nailed the beginning and made a mistake at the end. You can create a perfectly played comp using the beginning of the second take and the ending of the first take. You can use the same comping techniques to create a single musical passage from multiple musical ideas. As they improvise in the studio, musicians will often record a few takes and later comp the best ideas of each performance into a new, virtual performance.
Previewing the Takes Before you start comping, you need to become familiar with the takes you are going to comp. While doing so, you will assign the takes different colors to help distinguish between them, and then decide which part of which take you will use. The selected take folder and its takes fill the workspace. The take folder is on the Guitar track, and the three takes it contains are on lanes below the Guitar track. Take 3 at the top is selected and is the take currently playing.
The other takes are dimmed to indicate that they are muted. This is useful when you need to assign other regions the same color.
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