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My new Logic Pro Book is out! Great, David! Mac mini 3. Before this was limited. Legacy plug-ins, instruments, and channel strip settings can be accessed by Option-clicking each respective button.
Lgic packages - everything contained in one package instead of a folder still supports folder structure if desired. I also suggest the major features that were removed be documented directly. Especially prp fact that the bit bridge is now gone. Click the plus button on the channel strip as many times as you need to separate things I don't instrumnts that. First of all I reloaded a LP 9. Sounds like you're using the old key commands preset. Logic Pro X should have asked if you wanted to continue using the old ones or change to the new logic pro x instruments pdf free.
In the new ones the "T" toggles the tool logic pro x instruments pdf free. Here's a real time saver regarding flex on and off for tracks: just logic pro x instruments pdf free up and down! Logic X I hope it does it clip-based, because if thats the case this is some very nice integration right there. Record screen in quicktime, open. So no Apparently not, that option is missing from the bounce dialog. Another one for the "removed features" thread.
At the same time, even though I often used it, I can live without out it, although I really do not understand at all why they've removed it?
I cannot remember a logic pro x instruments pdf free person having issues because of that option. Mac mini 2. I really don't understand the removal of features. Logic pro x instruments pdf free they directly conflict with new software updates. Seems like теме!!! microsoft office professional plus 2013 freetrial product key free когда motto should be add rather than subtract. I actually like this better, to be honest. Although it would be nice go have some control of the file start and end points - now I assume lofic share you'll bounce anything between the project start- and end marker?
OK, I think everybody using "share to iTunes" should read this carefullybecause you will not immediately spot in the Share dialog if cycle mode is still on When you share a project to iTunes, the entire project, from the beginning to the end of the last region, is exported. Any silence at the beginning fref end of the project is trimmed.
I feel it should also show locators logic pro x instruments pdf free in 'normal' bouncing. But if I use U. Select onstruments command, click the Learn by Key Label button turns bluehit T. Hit Learn by Key Label button again, lkgic window, done. You can post now and logic pro x instruments pdf free later. If you fgee an loyic, sign in now to post with your account.
Paste as plain text instead. Only 75 instrumehts are allowed. Isntruments as a link instead. Clear editor. Pef or insert images from URL.
Click here! Logic Pro Share More sharing instrumenst Followers 0. Reply to this topic Start посмотреть еще topic. Recommended Posts. David Nahmani Posted July 22, Posted July 22, Link to comment Share on other sites More sharing options Jordi Torres Posted July 22, I'll keep adding to it as I find things to add.
Feel free to add to it too! Nice feature is that LPX will automatically try to assign a proper icon from sounds dragged from the loop browser, also from 3rd party content, probably by analyzing the naming of a file. Where can the discount coupons used toward the purchase of LPX be found? Logic Pro X Cool, I've added the ones you've contributed guys, thanks. Keep 'em coming! Posted July 23, Another thing that keeps being asked.
The default shortcut for the tool menu is "T. Mac OS X Pfd Nahmani Posted July 23, Thanks unstruments, logic pro x instruments pdf free those too. Great idea, David. Bass Amp Designer and new stompboxes in Pedalboard. Round robin sample support in EXS Alternatives integrated versioning.
Autosave in the background. Here's my tip for the day. With Drummer, you can have separate outs. Initiate drummer track 2. Jordi Torres Posted July 23, Chris D Posted July 23, Posted July 23, edited. Edited Olgic 23, by Chris D. Eriksimon Posted July 23, Cool, animated gif! How did you create that? Please say it's a freeware app Why did the chicken cross the Mobius ring? The ability to create ijstruments own color palette is gone in Logic X.
Ploki Posted July 23, FoolsGold Posted July 23, Is there a way to do Bounce to iTunes Посетить страницу Macbook Pro 2. I sorta liked it, saved having to go folder diving through the finder to find your bounce. Eric Cardenas Posted July 23, Nola Posted July 23, Sorry but back to the letter T for the tools list - I specifically said "Yes" the first time I started LPX when it asked me if I wanted to use the logiv set of key commands designed for LPX - but Esc is still what brings up the tool list for me, and T does nothing??
You know that can assign it it manually if you want? Type "tool" in its' search box optional, it's just for easier searching logic pro x instruments pdf free. Oh yes, I know it's all customizable.
Thanks for the tip. Join the conversation You can post now and register later. Reply to this topic Insert image from URL.
Logic pro x instruments pdf free -
Apparently not, that option is missing from the bounce dialog. Another one for the "removed features" thread. At the same time, even though I often used it, I can live without out it, although I really do not understand at all why they've removed it?
I cannot remember a single person having issues because of that option. Mac mini 2. I really don't understand the removal of features. Unless they directly conflict with new software updates. Seems like the motto should be add rather than subtract. I actually like this better, to be honest. Although it would be nice go have some control of the file start and end points - now I assume with share you'll bounce anything between the project start- and end marker?
OK, I think everybody using "share to iTunes" should read this carefully , because you will not immediately spot in the Share dialog if cycle mode is still on When you share a project to iTunes, the entire project, from the beginning to the end of the last region, is exported. Any silence at the beginning or end of the project is trimmed. I feel it should also show locators like in 'normal' bouncing.
But if I use U. Select the command, click the Learn by Key Label button turns blue , hit T. Hit Learn by Key Label button again, close window, done. You can post now and register later. If you have an account, sign in now to post with your account. Paste as plain text instead. Only 75 emoji are allowed. Display as a link instead. Clear editor. Upload or insert images from URL.
Click here! Logic Pro Share More sharing options Followers 0. Reply to this topic Start new topic. Recommended Posts. David Nahmani Posted July 22, Posted July 22, Link to comment Share on other sites More sharing options Jordi Torres Posted July 22, 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. Take 1 is purple. You will keep the blue color for Take 2, and choose a new color for Take 3. The selected take, Take 3, plays.
This time the first bar sounds good, but the second bar is rather messy; the third bar sounds good, and then the guitar player plays the wrong chord and stops. This time the guitarist misses the entire beginning but gives a good performance in the fourth bar of the breakdown. Although each take is a very poor performance, you have all the material you need to create a comp take that will sound good.
You will swipe your mouse across the parts of the takes you want to hear in your comp. The entire take is selected, and its color and name are displayed in the take folder. The mouse pointer does not automatically snap to the grid when Quick Swipe Comping, but snapping would help you edit this kind of rhythmic material. This time the mouse pointer snaps, making it easier to select exactly one measure. Notice that in the take folder on the Guitar track, the waveform and its background color match the sections of the selected takes.
Your comp name, Comp A, now appears next to the take folder name, and the letter A is displayed in the Take Folder pop-up menu to the right of the disclosure triangle.
An easy way to start a new comp is to Option-click a take to select it, and start comping again. There is, however, a lingering noise present at the end of Take 2 you can delete. The upper part of the clicked section is white, indicating that the section is selected.
You can hear a double-attack on the downbeat of bar You will now clean up that edit. Holding Control-Shift while you drag temporarily disables the snapping, giving you the precision you need to clean up this edit. The take folder is replaced by the current comp. The selected sections of the takes in the folder are now replaced by audio regions, and crossfades are displayed at the junctions between regions.
You now have a flawless funk rhythm guitar performance during the break. The crossfades, automatically added between edit points during the comping, ensure smooth transitions between the regions. You will learn how to apply and adjust your own fades and crossfades in the following two exercises. Adding Fades and Crossfades When editing audio, you usually want to avoid abrupt transitions on edit points: the region boundaries and the junctions between regions.
You can use nondestructive fades in the workspace to create smooth transitions. Adding a Fade-Out The very last region on the Guitar track ends abruptly, before the guitar chord has finished its natural decay. You will now add a fade-out to make that last chord end more naturally. You can hear odd blip sounds at the edit points: the beginning of the first region, the junctions between regions, and the end of the last region. The clicks are exacerbated by the reverb in the Amp Designer plug-in on the channel strip.
You can now clearly hear the clicks. The third region, a C minor chord, ends abruptly and the sustain tail of that chord does not sound natural. You can create fades only over region boundaries.
Here, the rectangular frame should cover the end of the region. A fade-out is created. The position where you started dragging determines the length of the fade-out.
The fade is curved in the direction you drag. The guitar and the piano fade out simultaneously at the end of the song, which now sounds cleaner and smoother. The Left-click tool is reassigned as the Pointer tool. Adding Fades to Remove Clicks In this exercise, you will add very short fades and crossfades to eliminate click sounds that occur at edit points on the final three regions on the Guitar track. You can hear a click at the beginning of the region. You may need to zoom in a few more times to clearly see the shape of the waveform.
To add fades using the Pointer tool, you can Control-Shift-drag over the region boundary. A fade-in is added. The click sound at the beginning of the Ab chord region disappeared. You can hear a click sound at the edit point. A crossfade is added at the junction between the two regions. The click sound at the junction between the regions disappeared. All you need is a very short fade at the edit point to smooth the transition.
This time you will add the crossfade using the parameters in the Region inspector to avoid zooming in and out. A five-millisecond fade-out is added at the end of the selected region. In the workspace, you can see that the fade-out at the end of the selected region is replaced by a crossfade.
After editing a section, you may have many small regions with fades between them. You can choose to keep those small regions with the fades so that you can readjust the edits later. However, if you are ready to commit and would rather deal with a single audio region for the entire section, you can join the regions to render your edits into a new audio file. An alert asks you to confirm the creation of a new audio file.
A new audio region is created in place of the selected regions and their fades. Zooming and scrolling in the workspace can help to an extent; however, when you want to edit the regions of a single track, you can use the Audio Track Editor to focus on that track without changing the zoom level of the Tracks area. Importing Audio Files Using the All Files Browser You will now import a new audio file to the project: a white noise sound effect you will use later to accentuate the transition between song sections at bar The All Files Browser opens.
At the top, three buttons allow you to access all the volumes connected to your computer, your home folder, or the current project folder. The contents of your home folder appear in the browser. The wave. A new track is created, and the wave audio region is added at bar The audio file was recorded at a low level, and its waveform is rather flat.
Depending on your zoom level, you may not even see a waveform at all. In the next exercise, you will zoom in to the waveform so you can see it clearly. The white noise effect sounds like it will work in that section. However, for maximum effect, it must be positioned so that the climax of the wave sound occurs at bar Using the Audio Track Editor You will now continue editing the wave region nondestructively, but this time in the Audio Track Editor, which allows you to clearly see the grid and the ruler above the regions without having to change the zoom level of the Tracks area.
The Audio Track Editor opens, displaying the wave track and its single region. The wave region fills the Audio Track Editor. You can clearly see the ruler just above the waveform, with vertical grid lines displayed under the waveform. You can see that the wave region is a stereo audio region because it has two interleaved circles next to its name, and two waveforms are displayed in the Audio Track Editor.
As you reach a certain zoom level, two waveforms are displayed, one for each channel. The waveform is a little taller. In the workspace, the wave audio region is moved accordingly. The climax of the wave sound is now perfectly aligned with the transition between song sections at bar The effect would sound even better if the rise before bar 17 were shorter. Then drag to the right so the region starts at bar The region is now trimmed.
All the edits you perform in the Audio Track Editor are reflected in the workspace. The wave sound now rises rapidly in the last bar of the breakdown and decays slowly in the next section, which works better for this transition. Playing an Audio Region Backward You will now create a new region from the last chord of the Gtr chords region at the end of the Guitar track, and copy it to the beginning of the song.
You will then reverse the new audio region to create a swelling sound effect during the introduction. You will now copy that region to bar 4, the last bar of the introduction. You have a new Gtr chords. In the Tracks area, you can see the Gtr chords. The swelling guitar chord sounds about right. To get the full impact of the break at the end of the intro, the Gtr chords.
To help line up the end of the reversed guitar with the first notes on the bass track, you can zoom in horizontally and position the playhead at the beginning of the Skyline Bass. Now the swelling guitar chord sounds smooth. Aligning Audio Accurately aligning audio material to the grid, or to other instruments in the song, is crucial to realizing a professional-sounding song. No amount of plugins, mixing, or mastering techniques can fix a sloppy arrangement, so getting a tight-sounding arrangement before moving on is important.
You will now import a guitar recording that was removed from the workspace but kept in the Project Audio Browser.
That guitar was removed because of timing issues, which you can now fix using the Flex tool. The third note, at bar 2, sounds out of place, while the other notes play at the second and fourth beat of each bar, much as a snare would be heard in a drum pattern.
You will move that third dead note to the second beat of bar 2. The audio files used on the Guitar track are analyzed for transients.
You may see a progress window briefly. You will learn more about flex editing in Lesson 7. Depending on its position over the waveform, the Flex tool can perform different functions, indicated by different tool icons. The dead notes in the first two bars now sound consistent.
The dead notes in this guitar region are still not located perfectly on the grid. If you wanted to take this a little further, you could set your snap mode to Beat, zoom in closer on the first guitar note, and use the Flex tool to drag it exactly on the beat. You now know how to read a waveform, identifying notes and their attacks to perform precise and clean edits.
You acquired skills with a number of editing tools—such as the Marquee tool, Fade tool, Resize tool, Flex tool, take folders, and snap modes—that you will continue to use as you edit recordings and arrange projects. Further, you can now accelerate your workflow by choosing the appropriate Left-click and Command-click tools for each job.
As you produce more music in Logic, you will continue sharpening those skills in the course of becoming an increasingly proficient audio engineer. What is nondestructive audio editing? Where can you perform nondestructive editing? How do you comp takes?
How do you prepare to edit the takes inside a take folder? How can you see the result of your comp as regions? How do you add a fade-in or fade-out to a region? How do you add a crossfade between two regions?
How do you select a section of an audio region? Which tool allows you to move an individual note inside an audio region without dividing the region? Audio region editing that does not alter the audio data in the referenced audio file 2. In the workspace or in the Audio Track Editor 3. Open the take folder, and drag over each take to highlight the desired sections. The take folder assembles a comp including all the highlighted sections.
From the Take Folder pop-up menu, choose Flatten. Drag the Fade tool over the boundaries of a region or Control-Shift-drag the Pointer tool , or adjust the Fade In parameter in the Region inspector. Drag the Fade tool over the junction of the regions or Control-Shift-drag the Pointer tool , or adjust the Fade Out parameter in the Region inspector. Use the Marquee tool. Goals Create a new project with a Drummer track Choose a drummer and drum kit Edit the drummer performance Arrange the song structure Edit performances in the new sections Customize the drum kit Tune and dampen individual kit pieces Work with electronic drummers Customize drum machines Convert Drummer regions to MIDI regions In most popular modern music genres, drums are the backbone of the instrumentation.
They provide the foundation for the tempo and groove of the piece. For recording sessions in which the instruments are not tracked at the same time, drums are usually recorded or programmed first so that the other musicians can record while listening to their rhythmic reference. In this lesson, you will produce virtual indie-rock, hip-hop, and electro-house drum tracks. Creating a Drummer Track Drummer is a Logic Pro X feature that allows you to produce drum tracks using a virtual drummer with its own personal playing style.
Its performance is placed in Drummer regions on a Drummer track. Using the Drummer Editor, you can edit the performance data contained in a Drummer region. Each virtual drummer also comes with its own drum kit software instrument plug-ins: Drum Kit Designer or Drum Machine Designer which controls Ultrabeat in the background.
A new project opens along with the New Tracks dialog. A Drummer track is created along with an eight-bar Drummer region. At the bottom of the main window, the Drummer Editor opens, allowing you to edit the performance in the Drummer region that is selected in the workspace.
The track is named SoCal Kyle , which is the name of the default drum kit and default virtual drummer in the Rock category. The project tempo is set to bpm, which suits the selected music genre. The drummer starts with a crash cymbal and plays a straightforward rock pattern. At the end of the Drummer region, a drum fill leads into the next section, which you will add later. If necessary, continue zooming vertically by dragging the vertical zoom slider or pressing Command-Down Arrow until you can see two lanes in the Drummer region.
The Drummer region displays drum hits as triangles on lanes, roughly emulating the look of drum hits on an audio waveform.
No analysis occurs when the Sampler analysis mode button is active. Click the Preview button to enable or disable automatic preview of selected files. Sound files can be mono or stereo, 8-, , , or bit, at any sample rate.
Note that rates above Click a filename, then click the Import button to import the sound. An import progress dialog is shown. The Import browser closes when the import is complete, and the previous window is displayed. Other file types are not supported for multifile import. If the note name is included in the filename, samples are mapped to corresponding keyboard zones for all analysis types.
When importing using additive or spectral analysis, samples without a note name are analyzed to determine pitch information that is then used for keyboard mapping.
Samples are mapped to the highest key of the zone and pitched down for the remaining notes in the zone. When no pitch information or note names are available, samples are mapped evenly across keyboard zones based on file selection order.
The root key pitch is set to the middle of each zone. Use standard modifier keys to select multiple files: Command-click to select or deselect files, Shift-click to select a range of files. If you are creating a sound that uses samples from multiple instruments or folders, drag the selected file or files to the Dropzone shown at the right side of the Import browser.
Once at least one file is added to the Dropzone, you can double-click a filename to add it to the list of Dropzone files. Drag filenames in the Dropzone list to change their order. You can use standard modifier keys to select or deselect files.
Choose a mode from the Mapping pop-up menu. Note: Clicking Import when at least one file is in the Dropzone imports files from the Dropzone, rather than files selected in the file list.
Logic Pro Alchemy source subpage controls Source components are shown only in advanced view. Each of the four sources has an identical set of controls for more in-depth editing.
Several source subpage controls are duplicates of those found in the source master controls. See Logic Pro Alchemy source master controls. For information on source filter controls, see Logic Pro Alchemy source filters. The waveform display and element controls section update to show relevant data and parameters when different synthesis engines are active.
The additive element is active for source A in the image. Note: If you save a preset with one source in solo mode, the resulting preset retains the on state of the source, but solo mode is disabled.
All other sources are off. If stereo mode is off and a stereo file is loaded or imported, only the left channel is played. When stereo mode is on, loaded or imported sounds are played in stereo. This makes it possible to pan individual oscillators in the additive element or individual grains in the granular element, for example.
See Logic Pro Alchemy source modulations. Loop Length determines the playback length, expressed as a percentage of the overall length of the sample. Speed has no effect when the element is set to sampler mode. Playback begins at the point set with the Position knob and travels through the audio data on a path determined by the Loop mode. Speed determines the rate of this travel. Playback remains at the normal pitch regardless of the rate of travel. Note: Global pitchbend behavior is determined with the PitchBend Up and Down controls in the master voice section.
See Logic Pro Alchemy master voice section. These positions are set with the Loop Start and Loop End knobs see entries above. Note off messages have no impact on loop repetitions. If the loop end point is placed before the loop start point, playback is reversed backwards to the loop end point when the loop start point is reached. The length of the loop, controlled with the Loop Length knob, is set as a percentage of the overall length of the sound, such that loop end point is equal to the loop start point plus loop length.
If the settings of Loop Start and Loop Length controls see entries above cause playback to go beyond the end of the underlying sample, silence is automatically inserted at the end of the sound. Note: The loop start and the loop end points, and sample start and end points, can be edited in the Main edit window.
The VA noise component is not affected by the Loop mode setting. The display also shows real-time additive or spectral data, or a representation of the waveform, if the Position or Speed controls are selected. When Position is adjusted, a position indicator is shown, allowing you to fine-tune the start position.
A variable combination of the Additive, Spectral, Pitch, Formant, Granular, Sampler, and VA buttons is available for use, depending on the audio data specified in the source select field and the import method used, if applicable.
Logic Pro Alchemy source filters Source components are shown only in advanced view. The source filter module provides three multimode filters, which can be configured either in series or in parallel. The source filters let you filter each source independently. The main filters, by comparison, process a mix of all four sources. Tip: Though you can use filters at multiple locations in the signal path, you can attain identical or similar results by careful use of fewer filters, which helps to reduce CPU load.
Source filter parameters The most common filter controls are listed. Note that the chosen filter type can change both the name and function of available controls, notably the Cutoff, Resonance, and Drive parameters. The three filters are independent and can have unique settings. The LED at the top of each button shows on lit or off unlit status. You can step through the available filter types with the Previous and Next buttons the arrows.
Series runs from filter 1 into filter 2. Parallel runs the two filters side-by-side. Resonance behavior varies among filter types. This can lead to intense distortions and aliasing, depending on filter type. Logic Pro Alchemy source filter use tips Alchemy provides multiple filter types at different positions in the signal path.
You can use filters at the source level, and the main filters, and you can also insert filters in the effects section. The positioning can have a significant bearing on the sound produced and can also impact on the processing resources required. Depending on currently available resources, you may need to pay close attention to envelope settings, the number of voices, and other parameters to achieve the sonic result you are chasing.
The downside is that source-level filtering requires more processing resources. Processing is per voice. Filtering at this stage of the signal path is often used to refine the overall sound or to provide a performance control variation. Use an FM filter as a sound source The FM filter produces a sine wave that is modulated by your source signal.
Because the FM process adds harmonics to the sound, the more complex the sound you feed into the filter and the louder it is the more quickly the sound is distorted.
FM in Alchemy is great for aggressive sounds, but is also useful for basses and other sounds. Unlike dedicated FM synths, Alchemy does not have preconfigured algorithms or a modulation matrix set up for FM synthesis. This means following the approach outlined in the steps below is not always the best option in Alchemy if you want to recreate classic digital FM sounds. Such sounds are often more easily achieved by other means, such as with additive synthesis or resynthesis.
FM in Alchemy is more like working with FM on analog synths where you modulate oscillator frequency rather than phase. In Logic Pro X, from the Name bar, click the File button and choose Initialize Preset from the pop-up menu to reset all Alchemy parameters to default settings.
Switch to advanced view, then click the A button to show source A parameters. Click the source A filter On button to enable the filter, then choose FM from the Filter type pop-up menu.
The filter is assigned to track keyboard pitch. By default, the centered knob at Hz provides a medium pitched sound that works well. If you want to change the octave, multiply or divide by two, and round to the nearest whole number that sounds best. For example, set the Frequency knob at Hz for one octave lower. Adjust the modulation Depth on the filter to increase the impact the source signal has on the sine wave generated by the FM filter, and listen to the results.
For more grit, try adjusting the Feedback control, which allows the filter output to apply modulation to itself. FM in Alchemy requires experimentation to develop more complex sounds. Here are a few things to try:. When doing so, resist high initial modulation and feedback depths so you can get a feel for the degree of control you have in shaping the overall sound. Important: Due to technical requirements, FM is often best done at the source filter level. As you progress through the signal path, gain increases and therefore FM become increasingly heavily modulated and distorted.
You will find that it is easier to work with FM at the source level than as a master filter or effect. Use a comb filter as a sound source When making comb filters your primary sound source, you may find that longer master envelope release settings are required for a natural feel when playing.
This results in more overlapping notes, higher polyphony, and therefore more CPU overhead. Because of this, you may need to carefully adjust envelope release times and reduce the maximum number of voices in the Master section.
By default, the centered knob at Hz will provide a medium pitched sound that works well. Set up an envelope for your impulse signal to excite the comb filter.
You can choose any impulse type, from VA noise to resynthesized samples. The aggressive nature of FM also makes this a good choice of impulse for comb filters. The impulse requires its own envelope, separate from the master or any envelope you have controlling the comb. The best settings for the envelope depend on the needs of the sound, but a good rule of thumb is to start with zero attack, zero hold, a very short decay, zero sustain, and zero release. This provides a quick spike that starts comb movement and allows the remainder of sound generation be handled by the comb itself.
Tip: The chosen impulse can have a large impact on the tone so it is worthwhile experimenting with different sound sources. One approach is to import a sample with a strong initial attack using the Additive import method, then use the Additive Harmonic effect knobs to adjust the tone. These controls plus comb filtering can provide numerous fast and easy adjustments, letting you dramatically alter the perceived hardness, material, and tone of your modeled sound.
You can also import a drum loop and set it to sustain with Continuous Loop mode. Because drum loops typically contain short bursts of sound that vary in tone, they work well with comb filters. Use the Damp control to reduce ringing or other artifacts in the sound, if required. Logic Pro Alchemy source elements overview Source components are shown only in advanced view. Each source can make use of multiple synthesis elements that operate on different functional principles.
You can use a synthesis method independently, or you can combine multiple synthesis types by turning on all required elements. For example, you can combine granular synthesis with virtual analog synthesis, or additive synthesis with spectral synthesis.
Each synthesis method has inherent strengths and weaknesses, making them more suitable for certain sound types than other synthesis engines. If you are new to synthesis or are unfamiliar with different synthesis approaches, see Synthesizer basics overview. Not all of these buttons are available for use at any one time.
The combination of active buttons updates to reflect the audio data specified in the source select field and the import method used, if applicable. See Logic Pro Alchemy Import browser.
Logic Pro Alchemy additive element controls Source components are shown only in advanced view. The parameters in this section are shown when the Additive button is active in a source subpage. The additive element controls also include a number of additive effects. See Logic Pro Alchemy additive element effects. Use Complex to choose a resynthesis waveform from the Shape pop-up menu.
Sine mode results in the most accurate resynthesis of the original sample and makes it easy to work with the additive effects and formant controls. In Complex mode, choosing any non-sine waveform can have a dramatic and often unusual effect on the overall timbre of the sound. Note: The additive effects and formant controls are named on the assumption that each partial is a sine wave.
To simplify working with these controls, it is recommended that the Sine option is chosen in most cases. When multiple elements are used in a source, use this control to set the relative level of the additive component.
The number of oscillators required depends on the sound. For example, a flute has a limited number of harmonics and requires fewer partials than a cello or a violin. The playable register can also affect the number of oscillators required: high notes will accommodate only a small number of higher harmonics before reaching the limits of audibility, while low bass notes may have hundreds of harmonics without reaching the limit.
Alchemy automatically sets a suitable Num Partials value when re-synthesizing additive data from imported audio files. You can reduce this value in some cases, but removing higher partials can cause unwanted changes to certain sounds. Note: The additive engine processes partials in groups of four. Set the Num Partials parameter to a multiple of four to achieve the best compromise between CPU load and sound quality. Always set Num Partials to the lowest number of partials that are required by the sound because this helps to reduce CPU load.
The sonic impact of this parameter is highly dependent on the audio material: sounds with strong inharmonic content such as bells are dramatically changed by reducing pitch variations.
If all partials are tuned to the harmonic series, however, the knob has no influence on the sound. The pitch variation knob is most useful when dealing with resynthesized audio. For example, if you analyze a vocal sample recorded with vibrato, this knob lets you reduce the vibrato depth, or remove it entirely with a setting of zero.
Removing all pitch variations from a vocal can result in a synthetic, artificial character. The audible effect is similar when the knob is turned in either direction. Symmetry alters waveforms until they are no longer pure sine waves in shape, resulting in each partial developing independent harmonics and making the sound brighter.
Click a source select field, then choose Import Audio from the pop-up menu. In the Import browser, click the Additive button to change the analysis mode. Single notes with a strong harmonic character tend to work well.
A good source of such files is the Vocals subfolder of the Factory samples. Play the file up and down the keyboard and reduce the value of the Num Partials knob to remove upper harmonics. Note that when playing higher notes you will need to turn this knob down further before you hear it start to take effect. Reduce the PVar knob value to remove all pitch variations and force all partials to a perfect harmonic series.
A PVar value of zero completely removes any vibrato or pitch changes in the source sound. Adjust the Sym knob value to change the symmetry of source sine waves, and note the extra brightness that is introduced by new harmonics that are generated. Logic Pro Alchemy additive element effects Source components are shown only in advanced view.
Three effects units are available in the lower half of the additive parameters shown in the source subpage. See Logic Pro Alchemy additive element controls. These are not audio effects in the traditional sense; rather they provide ways to control entire groups of partials simultaneously. Each unit provides a different selection of effects. Unit 1 is devoted to amplitude effects, unit 2 to pitch effects, and unit 3 to panning effects.
The initial configuration of the effects units changes when you use the default preset and when you import a sample. A sound with an imported sample loads the Harmonic module into unit 1. Note: Some effects are included only for compatibility with older Alchemy versions.
It is best to turn down your amplifier or mixer levels to avoid signal spikes that may damage your speakers or hearing. Your choice determines the controls that are shown. Choosing None results in an equal volume level for all partials, which can sound extremely bright and harsh. Set to zero to completely remove the fundamental tone.
Higher values tend to make the sound thicker. Set to zero to completely remove harmonics 1, 2, 4, 8, 16, and so on, while boosting the levels of non-octave harmonics.
Set to higher values to boost the levels of octave harmonics while reducing the levels of other harmonics. Low values increase the level of odd-numbered partials 1, 3, 5, 7, and so on , making the sound more hollow. Higher values boost harmonics 1, 3, 9, 27, and so on, with a corresponding reduction in the levels of other harmonics. Low values have the reverse effect and can make the sound more cutting and edgy.
Loaded by default when no sample has been imported into the additive element of the source. Turn toward Pulse to mute even harmonics. Low values increase the level of odd-numbered partials 1, 3, 5, 7, and so on, making the sound more hollow. The group of partials affected is defined by the Partial knob. The range of this knob is determined by the position of the Tuned knob.
Set to 2 to limit the detuning affect to only partials 2, 4, 6, 8, and so on; set to 3 to apply detuning to only partials 3, 6, 9, 12, and so on. Higher values affect fewer partials, which in turn changes the impact of the Amount knob, making it more subtle. This knob behaves like a switch. On: At the zero position, the selected partial is tuned down to the pitch of the second partial below.
Off: At the zero position, the selected partial is tuned down to the pitch of the second partial below. This stretching of the harmonic series is typical of instruments that use vibrating strings or tines. Higher values increase the intervals between partials and shift higher partials upward.
Lower values decrease the intervals between partials and shift higher partials downward. This is a quick way to tune upper partials without the need to manually edit partial pitch values in the additive editor.
See Logic Pro Alchemy additive edit window overview. Small increases can result in a subtle sweetening of the sound without altering its basic character. Larger increases can add an inharmonic, metallic, or bell-like quality to upper partials. Modulate this parameter with an envelope to add a plucked string type articulation to the start of a note.
In a sound with a fundamental frequency of Hz and a second harmonic an octave higher at Hz, an upward frequency shift of Hz results in partials at Hz and Hz, with the second partial no longer an octave higher than the first. This effect type can radically alter the harmonic structure, leading to inharmonicities and atonalities, in addition to a perceived change of the fundamental pitch. All subsequent partials are shifted by the same amount in hertz rather than in semitones because this would result in a simple pitch change.
Defining the frequency shift in this way retains a consistent timbre as you play up and down the keyboard. Because the shift does not track the keyboard, the timbre of the resulting sound changes from note to note. When combined with another harmonic sound in the VA section, for example , small shifts can create beating, chorus-type effects between the source elements. This can lead to unusual effects and can result in dramatic transformations of the sound. Subtle use of this parameter can turn harmonic pitched sounds into atonal bell-like timbres, for example.
All partials are shifted toward the target when the Amount knob is rotated. At low positions, partial pitches shift down. Adjust the knob to balance the shift and to control the brightness of any atonal, inharmonic elements that may be introduced to the sound. This can lead to chaotic effects. Partial knob: Set the amount that noise affects low or high partials by specifying the minimum partial that is altered by the effect.
This can create a subtle widening of the sound, with less obvious left to right movement in the lowest partials. At zero, all partials are in- phase. Higher values lead to a finer, more subtle and complex effect. The range is from 16 Hz to 20 kHz. Crossover frequencies are centered at Hz, Hz, and Hz, with a one octave transition between bands.
Partials are panned left to right in a regular pattern, with every second partial position inverted. Modulating the partial number can create rhythmic melodic effects. Shorter release times and slower modulation of the partial number can create strumming of partials. For example, a value of 10 applies a sine wave shape over partials , , , and so on. Modulate this with a ramp up LFO for smooth sound changes. Logic Pro Alchemy spectral element controls Source components are shown only in advanced view.
The parameters in this section are shown when the Spectral button is active in a source subpage. Two Logic Pro Alchemy spectral element effects units are available in the lower half of the spectral parameters shown in the source subpage. Note: You must first perform a sample import with a spectral analysis or draw in the Spectral edit window before you can use any of the spectral engine parameters. In Alchemy, the audible spectrum of a signal is split into a large number of spectral bins.
Energy distribution across these bins is analyzed and the sound is recreated by filling each spectral bin with the required amount of signal, using either sine waves or filtered noise. The results are then summed. Use Noise to fill spectral bins with filtered noise. The spectral bins are filled with sine waves, which is generally the best choice to recreate the entire original signal.
Noise mode can be useful for transforming normal speech into whispering, for example. The spectral engine is used only to recreate noisier aspects of the sound because this is not a strength of additive resynthesis.
In this case, the mode is set to Noise. When multiple elements are used in a source, use this control to set the relative level of the spectral component. All signals above this frequency are allowed to pass. Signals below the frequency are cut. 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.
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