n-Track Software
n-Track
Digital Audio Multitrack Recorder Software

USER GUIDE
REVISION 3.0.0
© 2011 Flavio Antonioli
n-Track Studio ©2011 Flavio Antonioli
Multiband Compressor, Graphic EQ and Tempo Delay plug-ins by Y. Oonisi.
n-Track Drums based on DK+ by Luigi Felici
Version 6.0 graphic design by Luca Panzarella
Manual
Portions written by Alessandro De Murtas.
v2 grammar revision by John Drummond.
v3 revision by James Podles
This manual, as well the software described in it, is furnished under license and may be used or copied only in accordance with the terms of such license. The content of this manual is furnished for informational use only, and is subject to change without notice.
We assume no responsibility or liability for any errors or inaccuracies that may appear in this book. No part of this publication may be copied, reproduced or otherwise transmitted or recorded, for any purpose, without prior written permission by Flavio Antonioli.
ALL PRODUCT AND COMPANY NAMES ARE TM OR ® TRADEMARKS OF THEIR RESPECTIVE OWNERS. WINDOWS 95, 98, ME, 2000 XP, Vista, NT and 7 ARE TRADEMARKS OF MICROSOFT CORPORATION.
Mac OSX is a trademark of Apple Computer Inc.
VST PLUGIN INTERFACE TECHNOLOGY BY STEINBERG SOFT- UND HARDWARE GMBH. ASIO ITERFACE TECHNOLOGY BY STEINBERG SOFT- UND HARDWARE GMBH.
Among the many people that have contributed in making the program always better with countless suggestions, comments, and bug reports, I’d like to thank in particular: Alessandro De Murtas, Sean Ercanbrack, J. David Lee, Jeff Keister, Richard Fairthorne, Lennard G. Cairns, Richard A. Smith, Michael Olsen, John Drummond, Bryan Bassett, Don Gaynor, David James, Kelly Craven, Tom Willis, Bax Taylor. Thanks to Ross Howard for the splash screen graphic.
Contents
1.1. Introduction To n-Track Studio
1.2. Installing n-Track Studio
1.2.3. Minimum System Requirements
1.4.3. Re-Recording Portions of a Track
1.4.4. Adding Effects and “Tweaking” a Track
1.4.5. Applying an Effect to Specific Parts of a Recording
1.4.6. Recording Using Markers
1.4.7. Final Mix-Down, Compression, EQ and Reverb
1.4.8. Programming Volume Envelopes
1.4.10. Mixing Down to a Single audio File
1.4.11. Editing the Mixdown File
2.1.1. Setting the Recording Levels
2.1.2. Signal Levels & Clipping
2.1.4. Audio Devices Selection Dialog Box
2.1.5. Recording More Than One Track at a Time
2.1.6. Recording Using Take Lanes
2.1.7. Punch-In/Multiple Takes Recording
2.1.8. Voice/Level-Activated Recording
2.2.1. Destructive audio Editing
2.2.2. Non-Destructive audio Editing
2.2.3. Editing Individual Samples
2.2.5. Snap Selection Edges To 0
2.3. Audio Effects and Signal Processing
2.3.4. DirectX Plug-Ins (Windows)
2.3.6. Effects That Work with n-Track Studio
2.3.8. Using other programs inside n-Track using ReWire
2.3.10. Aux Channels and Settings
2.3.12. Automating Effect Parameters
2.3.13. Volumes and Effects Parameters Automation
2.3.14. Aux Send/Return Automation
2.3.15. Destructive Processing.
2.3.21. Dither & Noise Shaping
2.4.1. Types of Files Used By n-Track Studio
2.4.2. Track Formats, Mono & Stereo Playback
2.4.3. Packed Song Files (.sgw)
2.4.4. Mixing Down the Final Song
2.5.2. Creating a Surround DVD
2.6.1. The n-Track Mixing Algorithm
2.6.2. 32-Bit and 64-Bit Versions
2.6.4. Fine-Tuning the Program to Your System
2.6.5. Customizing the Program
3.3. Creating a MIDI Drums Track
3.4. Controlling n-Track with MIDI Faders
3.5.1. VSTi/DXi Instruments Plug-Ins
3.5.2. Playing Live Through an Instrument Plug-In
3.5.3. MIDI Instruments Assignment
3.5.4. MIDI Instruments Definition
3.5.5. Programs (Instruments) Names
3.6. MIDI System Exclusive Messages
3.9.1. About SMPTE/MIDI Time Code
3.9.4. Configuring the Program to Act as Master
3.9.5. Configuring the Slave to an Incoming Time Code Signal
3.9.6. MTC/MIDI Clock Dialog Box
3.9.7. Virtual MIDI Patch Cables (Loopback Devices)
3.9.8. Syncing a Video Clip to Playback
4.5.1. Audio Devices (Advanced)
4.6.4. MIDI Events Editing Window
4.7. Additional Audio Settings
4.8.4. Sampling Frequency Conversion
4.8.5. Recovering a WAV File from Raw Audio Data
5.1. n-Track Studio On The Web
5.4. Creative Labs Soundcards Issues
6.1. Frequently Asked Questions (FAQ)
6.3. Upgrading To n-Track v6.x
For a long time, music recording has been difficult for those without access to professional recording studios. The inability to afford expensive recording hardware has confined many musicians to the world of second-rate equipment, discouraging their ambitions to realize their true musical potential. Once, this was typical, but today, things are changing! Forget private recording studios, expensive equipment and high invoices: with n-Track Studio, having a professional music home recording studio and a comprehensive audio environment has never been easier and more affordable!
n-Track Studio has been designed to bring you a powerful Digital Audio Multitrack Recorder. All you need is a full duplex soundcard, a microphone, some free space on your hard drive, and a bit of imagination: there are no limitations on what you can realize with this program!
Let’s have a look at some of n-Track Studio’s special features:
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Working with n-Track Studio is Easy |
Do it All by Yourself! |
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To start a new song, just record the first track (usually the rhythm track). To record a track, select either MIDI or audio recording by switching between the microphone and keyboard icons on the toolbar, then click the "Record" button. Once you've finished recording the track, click the "Stop" button. Can it be any simpler? |
Add and record as many tracks as you wish. Forget the track limitations of your old tape multitrack recorder: n-Track Studio can manage as many as thirty tracks on a high-end machine. |
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On-the-Fly Mixing |
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Instead of mixing audio tracks together, then playing them, n-Track Studio's mixing process is done "on-the-fly." This way, you can alter the volume and pan settings of individual tracks while listening to your song or even while recording.
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This feature allows you to start playing a track, then start recording at a specified point by clicking the "Record" button during playback.
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Live input processing allows you to use n-Track Studio as a multi-effect device for your instruments. You can connect an electric guitar to your computer, for example, then use n-Track Studio's effects as virtual guitar pedals. |
As with mixing, effects processing isn’t written on a file, but is calculated "on-the-fly" during playback. This lets you change an effect's parameters and listen to the results in real-time. The program comes bundled with many effects including EQ, Compression, Chorus, Echo, Reverb, Pitch shift, Tremolo and many others, and can also use third-party VST and DirectX (Windows) or AU (Mac) audio plug-ins. |
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Multiple Soundcards |
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n-Track supports multiple input and output soundcards, or more than one soundcard. This makes it possible to record more than one track at a time and to send the program's output to more than one stereo output. Each output channel has its own master effects section and master volume control.
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You can easily create drum and musical loops by concatenating several instances of the same audio file in a track. |
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n-Track Studio can sync to other programs or external devices using SMPTE/MTC or MIDI Clock sync, acting either as master or slave. AVI, MPEG and Quicktime videos can be played in sync with the song in a dedicated video playback window. |
In n-Track Studio, you can automate the envelopes for the volume, pan, send and return settings. This allows you to program fade-ins and fade-outs, crossfade between tracks, boost the volume of a track when there is a solo, and so on. |
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Integrated Drum Machine |
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Once you're finished recording the whole song, you can mix down all the tracks into a single audio file, then burn it to a CD. You can also use the built-in MP3 encoder to compress the song, then distribute it via the Internet. |
The built-in n-Track Drums module lets you easily compose drum tracks out of pre-built patterns and great acoustic or electronic drum sounds. |
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Download the installer file from www.ntrack.com, then double-click the downloaded file to start the installation.
Windows

Mac

Enjoy!
Windows

Mac

CPU:
Windows: Pentium III 700
Mac: Intel Core Duo
Memory: 512 MB RAM
Soundcard: A full duplex (i.e. capable of simultaneous playback and recording) soundcard
Operating System: Windows 2000, Windows Server 2003, Windows XP, Windows XP x64, Windows Vista, Windows Vista x64, Windows 7, Windows 7 x64
Mac OSX 10.5 (Leopard), 10.6 (Snow Leopard) and 10.7 (Lion).
Make sure you’ve got the latest versions of the drivers for all your hardware devices, especially the soundcard. Updating drivers is often the solution to many different problems. You can download updated drivers for your devices from the manufacturers' websites.
Working with n-Track Studio is easy. The first step is to check if the recording VU meter shows the input signal coming from your microphone or instrument. The VU meter is in the upper left section of the program screen, above the master volume control. If n-Track Studio is receiving an input signal, a green bar will appear in the VU meter. If it doesn’t, make sure that all the cables are connected correctly and that the soundcard is set to record from the correct sources. See setting the recording levels for more info.
Connect a recording source -- like a microphone -- to your soundcard's input jack. To start a new song, just click
the "Record" button on the lower toolbar to record the first track
(usually the rhythm track). Once you have finished recording, click the
"Stop" button. The track you've just recorded will appear on the
timeline as a waveform. Select it and adjust the volume
and pan settings by moving the sliders on the mixer window (you can do so while
listening to the track). Now you are ready to add a new track: make sure you
are ready to play and click the "Record" button again. Alternatively,
you can start the playback by pressing the "Play" button and then, during playback, clicking the "Record" button to start
recording at the desired point (this is called "punch-in recording"). Tracks can
contain more than one audio file, and you can drag and drop waveforms from one
track to another. You can also adjust the waveform's offset inside a track by moving
the small crosshair icon
in the lower left corner
of the waveform rectangle.
Once you have finished your song, save it as a .sng file using the Save As command from the File menu (File/Save As).

The .sng file that the program creates contains the audio files' names, the mixing settings, the volume envelopes and the effects settings. It does not contain the actual audio data, which will remain stored in the audio files as they were recorded. See types of files used by n-Track Studio for more info on the various file formats used by the program.
Now you may want to mix down all the
recorded tracks into a single audio file using the mixdown
command. Press Ctrl+R (Cmd+R on a Mac) to mix down the song, or select File/Mixdown
Song.
After you have successfully installed n-Track Studio, you can begin your first project with your new Digital Audio Multitrack Recorder. In this chapter, we will show you how to create a new song from scratch using just your computer, a microphone and n-Track Studio. Although the program is capable of recording two or more (depending on the audio hardware) tracks at the same time, for this tutorial we’ll assume that you’re going to record all the tracks by yourself, one at a time.
Prepare your instruments, turn on the computer, launch the program, and connect the mic plug to the soundcard’s mic input connector. Before you start recording, check if the program is receiving a signal from the microphone. Talk into the microphone and see if the program’s recording level VU meters move.

If they don’t, run the Windows volume control using n-Track Studio's Settings/Soundcard settings/Recording mixer controls menu command. You can also access this control through Windows: in Windows XP and earlier versions, select Start Menu/Programs/Accessories/Entertainment/Volume control. For Windows 7 and Vista, select Start Menu/Control Panel/Hardware and Sound/Manage Audio Devices. On a Mac, click the Apple menu, then select System Preferences/Sound.

Open the recording view (Options/Properties/Recording
in Windows; on a Mac, click the “Input” tab), then check that the mic in is the
only input source selected, and that the
level is sufficiently high.
While adjusting the mic input level, check the actual recording level using the n-Track recording VU meter. Sing or play the instrument the mic is going to record at the highest volume you think you will reach during the actual recording, and set the level so that, at the maximum volume, the VU meter will be in the higher red range.
Note: if you exceed the maximum allowed recording level, the recording VU meter will show a CLIP sign, which means the recording level in that precise instant has been too high and the recorded sound will be distorted.
Since
we’ve decided that we’re going to record one track at a time, and you’re
recording directly from the mic, you’ll want to record mono tracks. To set the
recording format to mono, click on the “Settings”
button
on the recording VU meter and select "Mono."
Since you’re recording your first track, there’s no particular need to use headphones, but you will need to use them for the following tracks to avoid feedback caused by the microphone capturing the signal coming out of the speakers.
Now that everything is set up, prepare yourself for recording and press the "Record" button on the lower n-Track toolbar. You’ll see the time indicator showing the recording time and a red flashing “Recording” sign. Complete the recording and click on the "Stop" button.
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Let’s say that you were a little nervous
about your first n-Track recording and the track didn’t come out right. No
problem: just double-click the recorded waveform, then click the
"Remove" icon
in the popup window that
opens. The program will ask you what you want to do with the recorded file. The
safest option to select is "Recycle Bin." This sends the file to the
system recycle bin, so that you’ll be able to retrieve it later if you realize
that you erroneously removed a good file.
Let’s say that you record the track again,
and this time everything goes well. You listen to it by clicking on the
playback button
on the toolbar, and it
sounds fine.

It is now a good time to save your work. By default, n-Track names the audio files that you record like this: “name_of_the_song_X.wav” (where X is a number). Since the default name of a project is “New song,” the WAV file you’ve just recorded will be named “New_song_1.wav”.
It's always a good thing to use a file name that is logically related to what the file is actually about, so we’ll now rename this file. Supposing that it is an acoustic guitar track, rename the file to "DemoSong_Ac_Guitar1.wav" using the Track/Rename-Copy wave file command. Type the new name into the "To" text box and click "OK." Notice that the name of the track, as shown in the upper part of the track’s mixer section, will not change: since a track can contain more than one file, the name of a track is not related to the name of its audio files. You can change a track name by right-click the track’s waveform in the timeline window, selecting "Properties" from the pop-up menu and typing a new name into the "Track name" box.
Save the whole project, using the Save as command from the File menu, to the file “DemoSong.sng”. The .sng project file will contain all the song settings, links to audio files, effects, etc., but not the actual audio data, which remains stored in the audio files that the .sng file points to, like the “DemoSong_Ac_Guitar1.wav” we previously recorded.
Let’s now record the second track. If you changed instruments, you’ll need to check the recording level again to be sure it’s correct. If you’re using headphones, you may also need to check if the audio being played back (i.e. the first track) is at a volume similar to that of the signal that you’re recording. This will allow you to play along with the recorded music while also hearing what you play through the headphones.
Setting up these levels properly may be a bit tricky with some soundcards: on certain soundcards, the recording level for the input source which is controlled by the slider in the recording view of the volume control/sound control panel, is not independent from the level of the input monitoring (the level in the playback view of the volume control/sound control panel). This means that you may not be able to move the mic monitor slider without altering the recording level. You can instead move the WAV out (playback) or master volume slider to match the playback volume to the volume of the microphone monitoring.
Once you’re ready, click at the point in the timeline at which you want to start recording the second track, then click the "Record" button. Record the audio for the second track, then click "Stop" when you're finished.

The most common problem at this stage is the so-called “bleeding” of the first track into the second track. If the soundcard mixer is not properly set up, the newly recorded track will re-record existing tracks, thus destroying the isolation between the tracks. To check if this is happening, just play back the tracks together and solo the second track by clicking on the S button on the track’s mixer section. If you still can hear the first track with the solo button pressed, your tracks are bleeding: see the Setting the recording levels section for instructions on how to solve this problem.
Let's assume that all went well and the second track (a vocal part) has been correctly recorded. Now you notice that the vocal level is a bit too low in some places and too loud in others. More often than not, vocal tracks need a bit of compression. Use the n-Track Compressor plug-in that comes bundled with n-Track Studio to flatten out the vocal's dynamics, making the quiet parts louder and the loud sections quieter.
Software effects processing via plug-ins is one of the most powerful features of software-based multitrack recording in general and of n-Track in particular. Effects plug-ins can be added in a number of different ways. By default, n-Track automatically adds a Compressor plug-in on each track channel. This is very handy because most channels need a bit of compression (and for those that don’t, simply leave the compressor at its default settings, which leave the signal unaffected).
Click on the "EQ Properties"
button on the track’s mixer strip next to the three small EQ knobs
. You’ll see the built-in EQ window appear and, below
it, the Compressor plug-in window.
Effects processing is always performed in real-time, so you can tweak the plug-in settings while listening to the result of the processing. Click on the drop-down preset list and choose a preset that is suited to the track (for example, “Soft knee compression”).
For detailed info on how to use the compressor, right-click on the plug-in’s logo and select Help from the popup menu.
Now that we’ve added a bit of punch to the vocal track, we’re ready for the third track: a couple of electric guitar solos, one in the middle of the song and one near the end. Record the third track as you did the other two.
Let’s say that the first guitar solo goes well, but the second doesn’t come out right. Since the first solo was OK, you don’t want to re-record the whole track, but you want to overdub the second solo.
Click on the
button on the left side of track 3, then select MME:
Microsoft Sound Mapper – Right Channel (or the name of your input soundcard
channel if your configuration is not the default).

Alternatively, you can click on the
button on the recording VU meter, then select "Record to
track 3" from the drop-down list.

Now click on the timeline at a point some seconds before the start of the 2nd solo, then click on the "Record" button. Once finished, you’ll notice that a new part appears on the timeline, just after the first solo audio file. In fact, when the program overdubs a track, it doesn’t modify the actual audio files: it just records another WAV file with the new material and adds it in the correct place in the timeline. This is called non-destructive overdubbing, as the original audio file is left unmodified.
The new recording is placed into a new Take; both the old and the new takes will appear stacked on top of each other. You can switch between the old and the new takes simply by clicking on the waveform of the take you want the track to play. Read more in the Take Lanes section.
Since we were doing this just to check what
had happened, we want to restore the parts exactly as they were after the
overdub. To revert the song to the previous state, click two times on the undo
button on the toolbar.

n-Track can also automate the process of recording takes of a portion of the song using punch-in multiple takes recording.
Adding Effects and “Tweaking” a TrackNow that the solos are OK, you may want to add a little EQ to the guitar track, and maybe also a bit of delay. Since other tracks may need to use this delay as well, put it on the first aux channel.
Click the "+" button in the guitar track's mixer section. When you hover the mouse over this button, “Add new send” appears. n-Track will add a new aux channel to the mixer and a send section at the bottom of the guitar track's mixer strip.
Right-click on the mixer window, then enable both the Show Master Channels and Show Aux Channels items. Right-click the mixer again, then enable the Horizontal Masters & Auxs item in the Layout submenu.
Move the mouse cursor over the black box underneath the Aux 1 mixer strip, then click the "+" sign. Double-click "n-Track" in the pop-up menu, then select n-Track Echo from the list of available effects.
The plug-in will now be applied on the first aux channel. To hear it applied to the 3rd track, you’ll need to send this track to the aux channel. Adjust the track send volume slider to vary the amount of signal sent from the track to the aux channel.

During playback, you’ll notice that the aux channel’s VU meter will start to move, but you still won’t be able to hear the effect of the delay. To make the aux channel's signal appear on the output, you’ll need to move the aux channel’s return slider, which is located directly underneath the master channel’s effect list. Once you’ve set it to an adequate level, you should be able to hear the delayed signal.

The Send mechanism is very flexible: any track or group channel can be sent to any other channel, allowing for complex and creative signal routing. To change the output of a channel’s send, click on the label at the bottom of the channel’s send section (the one that by default shows Aux 1 after the send is created).
Suppose you want the echo to be applied only to the final part of the second solo. We can obtain this using aux send (or return) automation:
· Add a new aux channel by selecting the Add new Aux channel command from the Add Channel menu (you don’t need to do this if the song already has an aux channel).
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Click on the volume
button on the toolbar and select “Draw send to aux channel
#1 volume” from the drop-down menu.
· Select the part of the waveform where you don’t want the echo to be heard by holding down the Alt (Control on a Mac) key and dragging with the mouse on the track.
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Click on the
button on the toolbar: you’ll see that the line that
represents the send volume envelope will be set to 0 (-Inf) for the duration of the selection. This
means that no signal will be sent to the aux channel during that time.

Start the playback and listen to how the transition sounds (to start the playback at a certain point, double-click on the time axis). If it sounds too sharp, soften it by substituting the abrupt step in the volume envelope with a short fade-in. Click and drag the points on the volume envelope to shape it into a gentle curve.
Now we’re ready for our last track: the
back-up vocals during the choruses. Let’s assume that the choruses' lyrics are
all a bit different from each other, and you don’t remember the song’s lyrics
too well. Use markers to mark the start of each chorus: click on the
button on the toolbar, and then click on the desired place on
the time axis. A triangle will appear to signal the marker's position.
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During playback, 5 seconds before n-Track reaches the marker, a pop-up window will appear. The pop-up shows the marker’s name, the count-down to the actual marker time and a text message. You can edit the pop-up name and text by double-clicking on the marker triangle on the time axis.
To help you remember the lyrics for the chorus, place markers at the beginning of each chorus, then type the lyrics of each chorus into the marker's text field. During recording, the lyrics will appear and you’ll no longer have the problem of remembering them (just like Karaoke).
All the tracks have now been recorded, so let’s do the final mixdown of the whole song. An almost mandatory step in the mixdown is to add compression to the whole mix by putting a compression plug-in on the master channel. Add the Compressor plug-in to the Master channel, then load the "Soft Limiter" preset and adjust the "Threshold," "Attack" and "Release" controls until your track sounds solid but not distorted.
A bit of EQ is also typically a good idea: add the Parametric EQ plug-in to the Master channel, then click and drag its equalization points to shape the sound of the track.
Add a reverb effect to the vocal track; if you want to use the same reverb on both the lead vocal track and the back-up vocals, put the reverb on the 2nd aux channel, then send both the lead vocal and the back-up vocal track to the 2nd aux channel. In this way, the reverb plug-in will process a single signal made up of the two tracks. This will use much fewer system resources (e.g. CPU power) than putting a separate reverb on each track, and will also make the two vocal tracks blend together better.
The next step in mixing is to refine the volume of the tracks during the evolution of the song. The volumes of the tracks don’t need to be constant: you can draw their envelopes in the same way we previously did with the aux send.
Click on the volume button
and select Draw volume envelopes from the drop-down
menu. Now you can design all of the track’s volume evolutions by clicking and
dragging the envelope points as you did with the aux send envelope.
We recorded an electric guitar solo track, and electric guitars are usually very noisy when they are not being played. You may want to remove the guitar buzz in the instants where the guitar is not playing.
Select the intervals to mute by holding
down the Alt key (Command on a Mac) and dragging with the mouse on the track.
Click on the
button to mute these parts. If the buzz is just at the start
and at the end of the audio files, you can obtain the same result by moving the
start and end of the waveform to where the guitar actually begins and ends
playing. To do so, drag the little rectangles at the left and right of the waveform
to the desired positions.
We’re now ready for the final fadeout.
Select "Master volume" on the popup menu that appears when you click
on the volume icon
on the toolbar. Select
the region where you want the fade to take place (typically the last few
seconds of the song) by dragging with the mouse on the time axis, then click
the fade out button
on the toolbar.
Now that everything sounds good, the last step is to mix down all
the tracks to a single audio file. Select the "Mixdown Song" command
from the File menu, then type in the destination filename and click
"Save." Click "More Options" to set the audio encoding
options. If you plan to burn the song to an audio CD, make sure that you check
the "Stereo" and "16 bit" options in the mixdown dialog box
and select "44100" as the sampling frequency.
Click the "Start" button to save the mixdown. The mixdown
process usually takes quite a bit less time than the actual duration of the
song. If you want to hear how the mixdown audio file sounds, save your project
(you should do this every time you make any important changes), then click the
button on the toolbar to remove all the tracks. Import the
mixed-down audio file using the Track/Import audio file menu command.
Listen to the track to hear how your mixdown sounds.
You may now need to cut unnecessary silent
lead-in and lead-out parts in the mixdown audio file. Click on the
button on the toolbar and select “Destructive audio
editing." Select the part of the track you want to cut out, then click on
the cut
button on the toolbar.
The last (optional) step is to burn the song to a CD-R. Open the CD burning dialog box using the View/CD Burning menu command. Import the mixdown audio file you just created, insert a blank CD and press the "Burn" button. You can also tell the program to mixdown and automatically burn the track to an audio CD by selecting the "Burn audio CD track" option in the mixdown dialog box.
Good luck with your next song!
This chapter contains information regarding the most common tasks that need to be performed when recording or mixing songs using n-Track Studio. Understanding these tasks (with particular regard to the interaction with the soundcard’s mixer) is strongly recommended in order to be able to use the program properly. To make your work with n-Track Studio as effective as possible, please take the time to read this chapter.
The first step in recording with the computer is learning how to set up the soundcard's recording software. Most soundcards contain a simple mixer circuit, through which the soundcard is able to select, among its many inputs and outputs, the signal(s) to record from and the signal(s) to send to the output. Before starting a recording, connect your audio source (microphone, guitar, mixer, etc.) to a soundcard input (usually marked "Line in" or
"Microphone in").
· Windows XP and earlier versions
Run the Windows Volume Control, which is usually
found in the Start Menu/Accessories/Entertainment menu. Choose the
"Options/Properties" menu command, select your soundcard and click on
the Recording button. In the options dialog box, make sure that all the
relevant signals are not hidden: the dialog box shows a list of the sources
that appear in the mixer, and sometimes important sources are hidden by
default. After you click on "OK," the mixer will show the view of all
the recording controls.
Make sure that the recording source to which you have connected your audio
source is activated (i.e. the checkbox below the source’s level slider is
checked).
· Windows 7/Vista
To adjust the audio devices settings, open the Windows Control Panel via Start
Menu/Control Panel. Select Hardware and Sound, then click on Manage
Audio Devices. Audio inputs (i.e. recording sources) appear in the Recording
tab of the dialog box; outputs appear in the Playback tab.
Each input and output of the audio device/soundcard will appear as a separate
icon with its own level meter. To adjust the settings of each input/output,
right-click on the input or output (for example, "Mic in"), select Properties
from the popup menu, then click on the Levels tab. The controls
available for each input or output may differ, and, depending on the particular
model of soundcard, there may or may not be controls for recording/playback
level, mic boost, etc. The Listen tab allows you to enable monitoring
the input signal by checking the Listen to this device checkbox.
· Mac OS X
Configure the level of the recording sources in the System Preferences ->
Sound panel. You can select which of the available inputs (i.e. line in or
mic) is the default recording device, and set the level of the device by moving
the Input Volume slider. The device you select for sound input is the
device that is used by n-Track when the Mac Default Recording Device input
device is selected in the Settings -> Audio Devices box.
Some advanced audio level settings can be configured in the Applications -> Utilities -> Audio MIDI Setup application. Some audio input devices allow monitoring of the input signal by enabling the Thru checkbox.
Turn off all unused recording sources to reduce the overall noise
level and improve the quality of the recording. Adjust the selected recording
source’s recording level slider, watching n-Track’s recording meters until
you obtain a good signal level without clipping.
You’re now ready to start your recording!
Note that hearing a signal coming from a soundcard input doesn’t necessarily mean that the soundcard is recording that signal: for example, most soundcards can be configured so that you can record the signal coming from the line input while monitoring (i.e. hearing audio from the soundcard’s output) the signal coming from the microphone input. You can set which signal to monitor from the Playback view/tab of the Volume Control/Sound Control Panel, while the signal from which to record is selected from the Recording view/tab.

Unlike in analog audio, where the recording level is somewhat
flexible, in digital audio, the audio recording level must always be lower
than the maximum possible level, usually indicated as 0.0 dB. Levels are
measured from the maximum level downward. For example, -90 dB is almost perfect
silence, -60 dB is barely audible and -6 dB is a very strong signal. 0 dB is the maximum, full-scale, signal level.
When the signal level goes above 0 dB, the signal is abruptly cut or "clipped," resulting in a very noticeable distortion called clipping. Unlike analog distortion, digital clipping is very unpleasant and should be avoided as much as possible.
For this reason, it's better for the signal to be a bit lower than the ideal level. If you use 15 of the 16 bits (i.e. use only half of the amplitude, peaking at -6 dB) you will probably not hear the difference (and you surely will not hear it if you use 23 bits out of 24), while if you sample a signal with an amplitude two times the maximum (+6 dB) you will probably hear bad clipping distortion.
The above is not true of the audio signals that flow within n-Track itself. It is true only for the signals that come from the soundcard (i.e. recording) and go out to the soundcard (i.e. playback output).
The signals inside n-Track's virtual mixer are floating point audio signals, which support levels far above 0 dB. Signals within the n-Track virtual mixer can reach very high levels without causing distortion, so you don’t have to worry that, for example, an effect is boosting the level of a track excessively.
The only thing that matters is that the overall master audio level,
as shown in the master channel level meter or in the Playback meter window
(View/ Playback VU meter menu command) is below 0 dB.
You can, for example, boost a track's volume so that the track level is +30 dB, as long as you reduce the master output volume (in this case, to -30 dB) so that the overall level is below 0 dB.
n-Track includes a soft clipping function, which tries to minimize the negative effects of digital clipping by automatically reducing the level of the signal when clipping is detected, and then gradually restoring the volume back to its original setting when clipping is no longer occurring.
While this works rather well, and avoids clipping distortion until the level is extremely high, soft clipping does modify the audio signal when reducing its level. This is much like what would happen if you had your hand on a stereo’s volume knob and adjusted the volume of a song very rapidly during playback to try to smooth out the loudest parts and boost the quietest parts.
Soft clipping can be disabled in the Clip sub-menu of the playback VU meter’s right-click menu.
The Audio Devices dialog box allows you to select which soundcards/audio interfaces the program uses for recording and playback. The dialog box can be opened with the Settings/Audio devices menu command
Selecting multiple playback and/or recording devices
To select multiple devices, click on the entries in the devices list holding the Shift or Ctrl key (on a Mac, hold the Cmd key). When clicking on an entry in the devices list, the entry will highlight and a number will appear to its left. The number indicates the order in which the program treats the devices. For example, using two recording devices, the 1st device’s level meters will be the leftmost ones in the level meters window. For output devices, the 1st device will be the device to which each track is sent by default. To send a track to a device other than the 1st, select the output device in the Track Properties dialog box that appears when you double-click on the track.
Audio Driver Standards
n-Track can access a soundcard using several methods:
· WaveRT: Introduced with Windows Vista, WaveRT is an audio driver
format
that’s specifically designed to enable audio applications to operate with very
low latency. WaveRT achieves a reduction in CPU usage at low buffering settings
by basically "getting out of the way" after the playback and
recording has been set up and letting n-Track talk directly to the audio
hardware without intervention by the driver. This translates to more efficient
streaming at low latencies.
Note: due to what appears to be a bug with Windows Vista x64, WaveRT
drivers allow you to use low buffering settings only when the x64 native
version of n-Track is used. When using the x86 32-bit version on the x64
version of Vista, low buffering settings result in an error when opening the
soundcard. Please see the 32-bit
and 64-bit versions section for info on the
difference between the 64-bit and 32-bit versions of n-Track
WaveRT drivers are currently available for Windows 7/Vista for a small number
of audio devices, including many motherboard audio interfaces such as the Realtek
ALC882M and C-Media 8738/8768.
· WDM: Introduced on Windows 98SE and supported by later
versions of Windows, WDM is comparable to Asio in terms of achievable latencies. Depending on your particular soundcard/audio interface, it might be best to use WDM, WaveRT or Asio drivers.
· MME: All soundcards have MME drivers. This is the oldest driver
standard and, although very reliable, it doesn’t allow you to work with low latencies and is consequently not suited for live input processing and playing instrument plug-ins live.
· Asio: Many semi-professional and professional soundcards have
Asio drivers, which allow for latencies similar to those of WDM drivers.
· CoreAudio: The audio device driver standard on Mac OS X.
Using WDM, WaveRT, CoreAudio or Asio drivers usually allows a much lower latency than other types of drivers, and makes it feasible to use the program for live input processing of a musical instrument played in real-time (for example, to use a distortion plug-in to play an electric guitar without an amplifier and process its sound within n-Track).
When using WDM, WaveRT or Asio drivers, selecting a single driver in the Audio Devices dialog box will typically enable all of the soundcard’s channels (as opposed to having to select multiple “virtual” drivers, one for each couple of channels, as you must with MME). You can limit the number of channels that you want to use in the Advanced dialog box.
When using Asio, the playback and recording buffers
size and number is decided by the soundcard’s driver. Changing
the settings in the n-Track Buffering settings dialog box will have no effect,
as the program will immediately restore the settings required by the Asio
driver. Many soundcards have a control panel that lets you change the buffering
settings; you can open this panel by clicking "Asio settings" in the
dialog box that appears when you click on a VU meter's "Settings"
button, then clicking "Asio Control Panel."
Windows Default Playback/Recording device is an alias for the device currently selected in the Control Panel/Sound-Multimedia applet as the preferred recording or playback device.
Suppose, for example that you have 2 soundcards, “X Audio” and “Y Technologies”, and the preferred audio playback audio device set in the Control Panel/Multimedia applet is “X Audio”. The content of the playback devices list will look something like this:
X Audio -> MME driver for X Audio card
WDM: X Audio -> WDM driver for X Audio card
Y Technologies -> MME driver for Y Technologies card
X Audio ASIO driver -> Asio driver for X Audio driver
WAV Mapper -> Alias for “X Audio”
In this example, the Y Technologies card doesn’t have an Asio driver.
The Advanced button in the Audio Devices dialog box opens the Advanced audio settings dialog box, which lets you adjust specific settings. The default settings should work well for most users.
n-Track supports simultaneous recording of multiple tracks. You can, for example, record the vocals and guitar simultaneously to two separate tracks, or even record a full band with each instrument being recorded to a separate track.
Regular soundcards/audio devices typically have one mono mic input and one stereo line input which can’t be used simultaneously; in this case, the maximum number of simultaneous tracks that can be recorded is 2. Dedicated multichannel audio recording hardware can have from 4 to 16 or more inputs.
To record two instruments playing together with a regular soundcard, connect one instrument to the left channel and the other to the right channel of the soundcard’s stereo line input connector. If the two instruments have line-level output (a guitar amplifier and a keyboard, for example), you can record directly into the soundcard. If one or both of the instruments are not line-level (microphones, for example), use a mixer to pre-amplify the signal and to route the instruments to one of the stereo channels using the mixer’s panning controls.
To enable recording the stereo line input to two separate mono tracks, set the Stereo to two mono tracks recording mode in the recording format dialog box.
When you re-record over an existing track, n-Track automatically creates a new Take for each recording attempt. Takes are a sort of “‘sub-track”; a track can contain multiple takes, and for each track you can tell n-Track to play one particular take, play audio from different takes for different sections of the track (i.e. play this chorus from take 2 and that verse from take 3), or even play all of the takes simultaneously (which can get pretty loud pretty fast!).
Say you have recorded a track and then recorded a new take on the same track. By default, the track will look like this:

When a track has more than one take, the track will be vertically split into lanes that show all of the takes available for a portion of a track stacked on top of each other.
The fun part comes when you want to the track to play a portion of one take and the rest from another take:
· Select the interval that you want to play from one take by horizontally dragging with the mouse over the track
· Press the “S” key (or select the Edit/Splice menu command)
· Now simply click on the takes that you want to play during the highlighted section

When a portion of a track has more than one take available, you can switch between takes by clicking on the take that you want to play during that portion of the track. In the screenshot, above the original take (Guitar.wav) plays for the whole length of the track except for the central part, in which the portion of take 2 (Lead1.wav) plays instead. Disabled takes will appear in a different color (purple in the screenshot above) from the take that is actually being played.
To switch between Take Lanes and the old way of showing and playing a single take per track, right-click a track, then open the Takes sub-menu of the menu that pops up and select “Merge Takes/Hide Lanes” or “Keep takes Separate.” You can also press the “L” key to cycle between views.
The Takes pop-up menu has also commands to move parts between takes, clone takes, and split a track with multiple takes into multiple tracks with one take each (which is handy when you want more control over the takes selection, as for comping).
After carefully listening to a solo you’ve just recorded, you decide that you’re still not satisfied, and you want to try recording it again. Click on the “Undo” button to send the audio file to the recycle bin. If you are not sure if you want to keep it, save an “alternate” version of the song before trying to re-record it, so that if you don’t manage to record another solo as good as the original one, you can always revert to the saved song.
After a couple of attempts, you start to get tired of having to use the mouse to start, stop, undo and restart the recording. Fortunately, you can have n-Track do this automatically for you using the punch-in function.
Select View/Toolbars/Punch-In-Multiple
takes recording settings/Show to make the punch-in toolbar appear. Click
the punch-in settings button
on the punch-in toolbar (the second button from the left):
![]()
Check the “multiple takes punch-in” option in the popup menu.
Drag with the mouse on the timeline to select the interval to record. Notice that as you drag, the punch-in toolbar is updated to reflect the selection.
Click on the start punch-in recording
button
to start the recording.
The program will let you repeatedly record the selected part. When you decide you’ve recorded a good take, click on the “Stop” button.
Each punch-in take is recorded to a new take on the same track. By default, n-Track shows takes on top of each other, allowing you to select bits from different takes by simply clicking on the take sections (see Take Lanes). If you find this confusing, and you prefer to view one take at a time, right-click on the track and select “Takes/Keep takes separate” from the popup menu. You can now switch between the takes by right-clicking on the track and selecting the desired take from the Takes popup sub-menu.
The punch-in toolbar allows you to set the
time at which the start and the end of the recording will occur. More options,
including the amount of time the playback will start before the actual
recording begins, can be specified in the punch-in settings drop-down menu that
appears when you click on the
button.
When you check the multiple takes option, the recording process will automatically restart when the cursor reaches the punch-in end time. The recording will continue to loop and accumulate takes until you click the “Stop” button. This can be useful, for example, when you want to record a difficult solo and you need to try it a few times. Using this option allows you to play continuously without having to manually stop the recording, remove the bad recording from the song and start again.
If the Add takes to new tracks option is selected, each take will end in a new track as soon as the take is finished.
The Voice/Level activated recording command in the Transport menu opens a box that lets you record only when the input channels have a level above a given threshold.
The feature is useful when doing very long recordings (for example, church services or audio surveillance) and you want to avoid recording long stretches of silence to save disk space and to simplify the subsequent editing and playback of the recordings.
The Voice/Level activated recording dialog box lets you set a Threshold level above which recording will occur, and the Hold time for which the program will keep recording when the signal goes below the threshold level. Use this hold time setting to avoid very frequent stops and restarts of the recording when the signal oscillates near the threshold level.

Editing Modes
n-Track Studio supports 2 different editing modes:
Destructive audio editing. In this editing mode, the program behaves like an audio editor.
This editing mode is called destructive because the files are actually modified.
Non-destructive audio
editing. In non-destructive editing mode,
n-Track performs copy, cut & paste operation using references to audio
files, rather than modifying the files themselves. You can also use this
editing mode to create
loops (to append the pasted part at the exact end
of the current track, hold the Shift key when pasting).
To change the current editing mode, click on the editing mode icon on the toolbar (which will show one of the icons above) and select the desired editing mode from the drop-down menu.
n-Track can also edit individual samples of audio files.
n-Track can perform basic audio editing functions, including cutting, copying, pasting, and inserting parts of audio files. To execute these operations, you must first select a part of an audiofile inside a track on which you wish to perform an operation.
Make sure that the arrow icon is selected on the toolbar and that the destructive audio editing button on the toolbar is enabled.
Holding down the left button, drag with the mouse on the desired audio file waveform to select a part of it. Alternatively, you can drag on the time axis at the top or bottom part of the timeline window. You’ll see the selection highlight as you drag. If you drag on the time axis, all the tracks will appear to be selected, but the audio editing operations will have effect only on the audio file that has the white border around its waveform.
Once the selection is made, click on the toolbar icon corresponding to the desired operation. Many operations will have no effect if the selection extends outside of the limits of the audio file, so make sure that the selected area is entirely within the audio file.
All destructive audio editing functions are undoable. When n-Track executes a destructive operation, it saves the data in temporary audio files to allow for multi-level undoing.
One useful destructive editing operation is to extract a part of a bigger audio file and place it in a separate track. To do this, select the desired part of the audio file, click on the “Copy” button and then click the “Paste” button while holding the Shift key. This will make the program create a new audio file containing the copied data. It will place the new audio at the exact offset that the copied audio had, so you can, for example, use this function to keep only the good part of a long recording take, deleting the original audio file, or to repeat a vocal part in multiple tracks offset by a small amount to create a choir effect.
Another useful trick is to silence parts of audio files. You can do this by selecting the part and clicking on the “X” button on the toolbar. If you hold the Ctrl key (or the Cmd key on a Mac), the operation will be destructive: the selected part of the file will be physically silenced. If you don’t hold Ctrl or Cmd, the volume envelope will be altered to mute the selected part of the track: to view the effect of this operation, switch to the volume drawing view by clicking on the “Volume” icon on the toolbar.
Very strange and cool effects can be obtained by reversing an audio recording. You can reverse the current selection using the Edit/Special/Reverse playback menu command.
For more sophisticated operations on audio files, n-Track can launch a user-specified (through the Settings/Preferences/Paths dialog box) external audio editor on the selected track. To launch the external editor, right-click a waveform and select “Launch external wave editor.”
In non-destructive audio editing mode, the cut, copy and paste function will never modify the audio files themselves. Non-destructive mode only modifies the parts that refer to the audio files.
Cutting a selection will have the effect of splitting the selected part into two pieces and shrinking them so that the selected range of the part’s audio file is no longer included in the song.
If the copy command is applied when the temporal selection is empty, the entire active part (the part you last clicked on) will be placed in the clipboard. To make the temporal selection empty, simply click on a part without moving the mouse.
Pasting a previously copied (or cut) selection will create a new part that exactly corresponds to the clipboard selection. If the paste command is executed when the current selection is not empty, the selection will be filled with the clipboard part, while if no selection is active, a new part will be created in a new track.
Editing Shortcuts
· Shift+Ctrl+V: holding the Shift key while pressing Ctrl and V will append the part currently in the clipboard (i.e. the part which has been previously copied or cut) at the exact end of the current track (see looping audio files). On a Mac, use Shift+Cmd+V for this shortcut.
· Shift+Ctrl+X: cuts out the selected section of the track or whole song, shifting the remaining portion of the song to the left. This shortcut is useful when cutting out the silence at the beginning of a song. Mac users can run this shortcut by pressing Shift+Cmd+X.
Other non-destructive operations are
accessed through the Edit menu:
· Splice: detach the current selection from the rest of the audio files; a reference to a audio file is cut into 2 or 3 pieces. The Splice command actually works in two modes:
When a track selection exists (i.e. a portion of one or more tracks is highlighted), the splice command works on the current selection. It doesn't look at the position of the vertical playback cursor; i.e., if you select a portion of a track by dragging with the mouse over the waveform, then single-click on a different position in the timeline axis, the Splice command will be applied to the selection you dragged on the waveform, not at the point at which the cursor is positioned. The selection will be transformed into a new part, thus creating 3 parts from the original part.
When no track selection exists (for example, if you press
Ctrl+0 or Cmd+0 to clear all selections) the Splice command will be applied at
the timeline cursor position, and it will split the current part into two
parts, one before and one after the cursor position.
· Splice in N parts: Creates N concatenated parts out of the selected part. Setting this command to “5,” for example, splices the selection into five equal portions.
· Merge: Rejoins two parts that have been previously spliced.
Audio files are made up of a series of samples. Each digital sample is simply a value between -1 and 1. The sampling rate at which an audio file is recorded represents the number of samples that the audio file contains in each second of recording. When you look at a waveform, you can’t usually discern each sample, as, even in short waveforms, the number of samples is very high, and they are “blurred” together to form the waveform representation that you see on the n-Track timeline.
If, however, you zoom into the X axis by
repeatedly clicking the
button
on the toolbar (or by rotating the mouse wheel forward while holding the Ctrl or
Cmd key), at a certain point, you’ll start to see a series of dots appearing on
top of the waveform, as shown in this screenshot.

The dots represent the individual samples that make up the audio file. You can drag a dot vertically with the mouse to adjust the sample value.
The ability to edit individual samples can be useful for correcting DC offset problems in recordings and for harmonizing editing points. If, for example, you place two audio files next to each other and, during playback, you hear a short click corresponding to the point where the two audio files are attached, you might be able to eliminate the click by editing the samples near the connection point to make the transition visually smooth. An alternative method is to overlap the two audio files slightly and cross-fade the two.
Note that abrupt changes between close samples do not usually appear in real recordings. Manual edits of individual samples that result in abrupt changes in the sample value (the screenshot above shows such an abrupt change) results in very high-frequency noise bursts that can be very annoying and hurt your ears.
Normalization is the process of amplifying
an audio signal so that its maximum amplitude matches a specified level.
Normalization can be useful, for example, when preparing an audio file for
burning a CD. Setting the maximum level of all CD tracks to 0 dB assures that
no clipping occurs and that the playback level of all tracks is similar
(assuming that all the tracks have been processed with similar compression and
limiting settings).
To normalize a waveform, open the Edit menu and select “Normalize.” Configure the options in the Normalize dialog box to control the normalization; to expand the controls, click the “More Options” button.
· Normalize to: sets the maximum level that the signal will assume after normalization
· Scan: scans the file to extract the current maximum level
· Channels: selects which channel to apply the normalization to. If the normalization is applied to both channels, the amplifying factor will be chosen so that the channel which has the highest peak will reach the requested level. For example, normalizing to 0 dB with a stereo file whose left channel peaks at –3 dB and whose right channel peaks at –2 dB will produce a file in which the left and right channels peak at –1 and 0 dB, respectively. This option only appears when a stereo waveform is selected.
· Apply to:
o
Selection: processes the selected
portion of the audio file
o Whole file: processes the whole audio file
· Convert to:
o Stereo: converts a (mono) audio file to stereo
o 32-bit: converts the file to 32-bit format
· Dither: Enables dithering.
· Dither depth: Sets the depth, in bits, of the dithering noise.
· Noise shaping: Enables noise shaping.
Selecting good cut and paste points is crucial to obtaining natural-sounding edits of audio files. The Snap to 0 option is designed to help the selection of such points: the instances in which the waveform crosses the 0 level are often the best places at which to cut an audio file. Enable Snap to 0 in the Edit menu.
You can adjust the Snap to 0 settings by opening Settings/Preferences/Options and clicking the “Snap to 0 settings” button:
· Snap when crossing 0 with [Negative/Positive/Any] slope: the program can detect the slope of the crossing of the 0 level. It's usually good to use a positive or negative slope; this way, when a segment of audio is pasted into another, the slope of the resulting waveform at the insertion point will be sufficiently smooth. This reduces the appearance of clicks at edit points.
· Scan at most [x] samples: sets the number of samples that the program will scan when searching for the 0 crossing.
· Assume DC component: sets the signal level that the program will consider as 0. This option is useful if the file you're working on has a DC component: the snapping will be made relative to the DC value specified rather than relative to 0. The level must be entered as a real number between 0 and 1 (e.g. “0.23”).
Crossfading creates a smooth transition between two separate audio files. n-Track performs crossfades in real time during the playback of a song.
This operation is non-destructive because the original audio files are not modified.
To apply a crossfade, drag one edge of a waveform so that it overlaps another waveform by the desired amount (the crossfade time). As you drag, you will see the crossfade volume envelope shape appear in the space at the intersection of the two waveforms.

You can disable the crossfading of a audio file in the Crossfade sub-menu of the popup menu that appears when you right-click on the crossfade area at the intersection of the waveforms. Select “More” from the popup menu to customize the length and shape of the crossfade volume envelope.
To loop an audio file in n-Track Studio, do the following:
· Insert the audio file you want to loop using the Track/Insert audio file menu command
· Make sure you’re in non-destructive editing mode (click on the editing mode button on the toolbar)
· Press Ctrl+C (or select Edit/Copy). On a Mac, press Cmd+C.
· Hold down the Shift key and press Ctrl+V (Cmd+C on a Mac)
This will insert the same audio file again. When the Shift key is pressed and held, the program automatically puts the new reference to the audio file in the same track as the copied part with the offset equal to the former end of the track, so that there will be no gap in the playback. Continue pressing Shift+Ctrl+V (or Shift+Cmd+V on a Mac) until the audio file is repeated as many times as you wish.

You can also use this technique to create more complex loops. For example, if you want to create a drum track and you have two audio files, one for the normal bar and one for a break, you could paste the normal bar 3 times, the break bar 1 time, then copy and paste the whole sequence several times to make a long drum loop.
Markers can be inserted by clicking on the
button on the toolbar, and then clicking on the desired place on
the timeline’s upper time axis. You can also insert a marker by right-clicking
on the time axis and selecting “Add marker here” from the pop-up menu.
To move a marker, simply drag it on the timeline axis. Double-clicking on a marker will open the marker properties dialog box, in which you can change the marker’s name, position, comment, and pop-up text.
Hold the Ctrl key (on a Mac, hold the Cmd key) and click between two markers on the timeline axis to set the edit selection between the two markers. Use Ctrl+Shift (or Cmd+Shift) with the left and right arrow keys to move between markers.
If a marker’s pop-up text is not empty, during playback, 5 seconds before the
marker time, a pop-up window will appear. The window shows the pop-up text and
a countdown of the time remaining before the marker’s position. This feature
may be useful, for example, during recording: you could let the program warn
you just before some critical points of a song are approaching (for example
“first guitar solo”, “last chorus,” etc.).
The Save/Recall selections window allows you to save and recall the selections used for editing on the timeline window. Saving and recalling selections is a method complementary to using markers to define edit points. Access this window by right-clicking on the timeline and selecting “Save/Recall selections.”
· Add: Save the current selection
· Delete: Delete the selection highlighted in the list of selections
· Merge: Merge the selections highlighted in the list of selections together
· Apply: Recall the highlighted selection
· Close: Close the window
n-Track allows you to define regions within audio files. To define a new region, highlight the section of the audio file that you want to be included in the region by dragging with the mouse on the timeline window, then right-click and select “Create audio file region” in the pop-up menu.
You can find the list of all the loaded audio files regions in the View/audio files regions dialog box. Add a part made of a selected region to an existing or new track by dragging the desired region from the regions list dialog box to the timeline window.
The information on an audio file’s regions is saved in the audio file itself, in a format compatible with most Windows audio file editors.
· Insert part at 0: Add the selected region to the beginning of a new track
· Insert part at original offset: Add the selected region to a new track starting at its original offset position
· Delete region: Delete the selected region
· Show regions in waveforms: Check this option if you want the regions to be shown in the waveforms representing the song’s audio files in the timeline window.
· Close: Close the window
Realtime audio effects can be used from within the program in a number of ways:
· Insert effects
· Aux channel effects
· Master channels
Insert effects process the signal before it’s fed to the main mix. Up to 25 effects can be applied to each track. To apply a new insert effect to a track, click the “+” icon in the track’s inserts effects list in the mixer window, then select the effect from the window that pops up.
Aux channel effects allow you to send the signals from multiple tracks to one channel, then apply one set of effects to the mixed signals. After processing the signal, the aux channel feeds it to the main mix. This kind of effects processing is extremely useful for certain kind of effects, especially reverbs and delays. Instead of applying a reverb to several tracks using insert effects, you can put a reverb on an aux channel, then use the send control of each track in combination with the aux return control to adjust the amount of reverb applied to the track. To add an effect to an aux channel, click the “+” icon on the channel’s effects list in the master mixer window. Select an effect from the drop-down menu. Each track signal can be sent to an aux channel using the track's send controls, located below the list of inserts effects on the mixer window.
Use send automation (drawing the send evolution in the timeline) to apply effects to certain parts of a track, instead of applying them to the whole track, or to vary the amount of an effect during the course of a song.
At the end of the signal path, the master channel effects process the signal resulting from the mixing of all the tracks and aux channels (if needed, you exclude aux channels from the master channel effects; see aux channels settings). The master channel effects chain typically includes an EQ plug-in, as well as a compressor and/or a limiter, which allows you to obtain a good volume impression without clipping.
Effects can be arranged in any combination, and one single effect type can be repeated several times for a single track (for example, you can add two separate echoes with different delay times to a track). To change the parameters of a particular effect on a track, open the effects dialog box and select the desired effect in the track list box. The effect dialog box will appear and you will be able to make the desired changes. You can alter the order in which the effects are applied using the up and down arrow buttons.
When using effects, be aware that the program calculates them while the song is being played back, so the load on the computer processor increases quite heavily. On the other hand, effects are non-destructive, meaning that the track that the program processes isn’t really modified. This allows you to experiment without having to worry about ruining the audio files and, more importantly, without having to wait for an audio editor to process the whole track.
You can make effect processing permanent by destructively processing a track with its current track effects.
Decrease the load on the CPU by freezing
tracks, instrument and group channels.
n-Track Studio allows the computer to be used as a multi-effect device. This feature allows you to connect an electric guitar, for example, to your computer, then use the program’s effects as virtual guitar pedals.
To enable Live Input Processing:
· Click the Live button on the main playback/record toolbar.
· Add a blank audio track, click on the small Record button in the track's section of the left timeline bar, select the input you want to receive audio from, then make sure that the Monitor Live Input button next to the small record button is active. You should now see the track's level meter move according to the input signal.

When in live input processing mode, the program will record from the selected soundcard input(s), feed the recorded signal into the main mixer, running it through any enabled effects, then output the resulting signal to the active output soundcard(s).
If the buffer sizes are sufficiently low, the input to output latency will be low enough for the processing to appear to the ear as if it’s done instantly. This enables applying effects to an instrument while playing it through the computer. You can, for example, add a delay or reverb to an electric guitar, or add reverb to vocals.
Live input processing also works during song playback and recording. This may be useful for monitoring the tracks being recorded. Vocal tracks, for example, are often processed with compressors and reverbs. Normally one would record the track dry (without any effects), then apply the effects. The drawback with this way of recording is that, during the recording, the performer doesn’t hear how his or her voice will sound when processed with the effects. Using live input processing, you can add the desired effects before recording the vocal, then record the track with live processing enabled so that the program processes the voice with the effects and mixes it with the other tracks in the song.
You may notice a small sound delay when using live input processing. This is due to the audio buffering in the soundcard. Carefully adjusting the buffer settings in the buffering settings dialog box is fundamentally important for obtaining the lowest possible input-to-output intrinsic delay.
Turning the buffering knob to the left will decrease the total buffering, thus allowing for a smaller delay: if the buffering is insufficient, a slight distortion (actually a fast repetition of clicks) will be hearable.
To make the intrinsic delay less annoying, only listen to your instrument through the output of the computer. For example, when playing an electric guitar, connect the amplifier output using its line-out jack, then mute its speaker output (by plugging a headphone adapter into the headphone connector, for example).
If the intrinsic delay is so high that you can’t manage to play while hearing the delayed sound, adjust your setup so that you hear your instrument output from both the computer speakers and from the instrument amplifier speakers (if applicable), even if some effects, such as EQ or compression, lose their usefulness.
The minimum delay of a system doesn’t
depend much on the computer speed, but mostly on
the soundcard’s driver design: the best results are typically
obtained when using Asio, WDM or WaveRT (Windows) or
CoreAudio (Mac) soundcard drivers. When input to output latency (the sum of
the input latency and the output latency) falls below 10 milliseconds, the
delay becomes un-hearable to most people.
n-Track Studio supports VST 2.x and 3.x standard plug-ins.
You can apply a VST effect to a track by right-clicking on the track and selecting Effects from the pop-up menu. Each track or channel’s mixer stripe contains a black list box that lists the plug-ins currently applied to the channel.
The property window for each plug-in contains the bypass checkbox, the CPU time indicator (which shows the percentage of the total program time used by the plug-in) and the preamp/postamp controls. These controls allow you to adjust the level of the signal arriving to the effect so that no distortion is introduced and that the signal level is kept sufficiently high.
If you don’t need these adjustments, keep these controls in their central position (an “off” label will appear below them) so that the program doesn’t waste CPU time applying inaudible amplification. To quickly turn off one of these controls, right-click it and choose “Center.”
VST 2.x plug-ins must be stored in a single
folder on the hard disk. You can specify the path to this folder in the Preferences/Paths dialog box. The location of VST 3.x plug-ins
is standardized, so you shouldn’t typically need to worry about the VST 3.x
plug-ins location: the plug-ins’ installers should take care of that.
n-Track Studio supports the DirectX standard plug-in architecture.
Some DirectX plug-ins will refuse to work with mono tracks, reporting an error when you chose them from the list. If you need to work with one of these plug-ins, you can either use it only on stereo tracks, or check the option for “expand mono tracks to stereo” in the Preferences/General dialog box. This will also greatly enhance the result of some effects, in particular reverbs.
The property window for each plug-in contains the bypass checkbox, the CPU time indicator (which shows the percentage of the total program time used by the plug-in) and the preamp/postamp controls. These controls allow you to adjust the level of the signal arriving to the effect so that no distortion is introduced and that the signal level is kept sufficiently high.
If you don’t need these adjustments, keep these controls in their central position (an “off” label will appear below them) so that the program doesn’t waste CPU time applying inaudible amplification. To quickly turn off one of these controls, right-click it and choose “Center.”
The effects list may contain some entries that aren't actual
filters, but are in fact Windows internal codecs. Typically, if you choose one
of them, a pop-up dialog will appear saying "Filter doesn't support
property pages.”
n-Track Studio supports the Apple AU (Audio-Unit) plug-in standard.
Your Mac comes with many Apple AU plug-ins, including Reverb and Compression units, and the default MIDI output on Mac is the Apple AUi DLS instrument, which provides General MIDI instruments sounds.
You can find links to shareware and freeware VST, DirectX and AU effects plug-in on the n-Track Studio website.
Effects processing can sometimes overload the CPU (the computer’s “brain”), especially when working with complex projects or when the computer is not very fast, resulting in annoying clicks or pops while playing the song. One way to overcome this problem is to Freeze channels that are being processed by CPU intensive plug-ins. When you freeze a channel, n-Track creates a temporary WAV file with the channel’s audio data, including any effects plug-in processing. During subsequent playbacks, instead of processing the channel with the plug-ins, n-Track simply reads the temporary WAV file you created when you froze the channel.
To freeze a track, select Track/Freeze/Freeze Track. When a channel is frozen, the effects are bypassed, and changing an effect’s parameters wont’ have any effect until you de-freeze the channel by selecting Track/Freeze/Unfreeze Track. If you de-freeze a channel and want to freeze it again, you have two options: either repeat the regular freeze procedure just like you did the first time, or use the Re-Freeze command, which instructs n-Track to simply re-use the temporary WAV file created during the earlier freeze. Re-freeze is instantaneous, although it won’t reflect any changes in effects parameters done after the original Freeze.
Freezing is also available on instrument channels: if a VST or DX instrument is starting to use too much CPU, you can simply freeze the channel, and the MIDI tracks sent to the instrument will be frozen too. You’ll still be able to hear new MIDI tracks sent to the frozen instrument or MIDI notes coming from an external MIDI keyboard.
The Bounce command consolidates tracks made of multiple audio files into a track with a single audio file. To bounce a track, right-click it, then select “Freeze/Bounce” and “Bounce to single wave file.”
The Bounce and Process track command in the Freeze/Bounce submenu consolidates the track into a single audio file, and also gives you the option to permanently apply track effects, EQ, and volume envelopes. It also gives the option to create a audio file that starts at the beginning of the song for easier exporting of tracks to 3rd-party programs (see Tips below).
Tips
· Hold down the Ctrl key (Cmd on a Mac) while running the Bounce command to bounce all the tracks in the song.
·
When you launch the Bounce and Process track command, the
Bounce Options dialog box will appear. Select Bounce from beginning of song and
click on Bounce all tracks. This will bounce all the song’s tracks so
that you’ll be able to easily import the tracks into a new song, automatically
keeping them in sync with each other. This command is also useful for preparing
the tracks for exporting the song to 3rd-party
multitrack editing programs.
n-Track Studio can host ReWire-compatible third party software such as Propellerhead Rebirth, Reason, Ableton Live, Virtual Sampler and more. A ReWire-compatible slave program will automatically send its signal into the corresponding n-Track ReWire channel(s). You’ll then be able to process the signal with plug-ins, send it to aux channels, group channels etc. ReWire software will sync to n-Track’s timing, and the transport controls (play, rewind, etc.) on the ReWired software will work together with n-Track. Starting the playback from a Rebirth window, for example, will also start the n-Track song’s playback.
To add a ReWire channel, select Add Channel/Add Rewire Device. Select a program from the list. A window with the available channels will appear. A ReWire program can have two or more channels; by default, n-Track activates the first two channels. Other channels can be activated “on the fly” during playback. Once you’ve added the ReWire channel, you can load the ReWire application and test if its signal is correctly being routed through n-Track.
Once you’ve finished your work, close the ReWire application first, then n-Track. Closing the programs in the reverse order may cause stability problems.
ReWire technology is developed by Propellerhead Software AB.
Side-chaining is a mechanism by which an effect plug-in can process the signal of an audio track based on the characteristics of the signal of a different track. A typical use of side-chaining is when the dynamics (i.e. compression) of a bass track is altered based on the dynamics of a kick drum track, or when a radio speaker talks over music being played, with the music automatically decreasing in volume when the speaker talks (“ducking”).
A plug-in needs to have explicit support for side chaining: for example, many compressor/limiter plug-ins have a switch that lets them use a sidechain or external input as a key signal.
Compressor plug-ins that have support for side-chaining include DensitymkII (free), Voxengo Crunchessor, and FabFilter Pro-C (available as VST3).
To use side-chaining with n-Track:
· Add a plug-in that supports side-chaining to the controlled track. This is the track you want to alter (the bass track, for example).
· Click on the output label in the mixer stripe of the track that you want to use as the side-chain (the controller track, which is often the kick drum track).
· Select Add new send from the pop-up menu.
· Click on the send output label of the send that you’ve just created. The output selection pop-up menu should list the side-chain input of the plug-in in the controlled track. Select this menu entry.
· Set the option in the side-chain plug-in to activate the side-chain input, if required by the plug-in.

If you only want to use the controller track for the
side-chain source, without actually hearing any audio from the track, send the
track’s main output to the plugin’s side-chain input instead of using a send
output.
To send a track to an aux channel:
· Click on the output label in the mixer stripe of the track that you want to send to the aux channel.
· Select Add new send from the pop-up menu.
· Click on the send output label of the send that you’ve just created.
The output selection pop-up menu should list the available aux channels (Aux channel 1 will be automatically created if it was not previously present in the song). The newly created aux channel should appear in the Master Mixer window (View/Mixer/Master mixer menu command). The aux return section for the aux channel you just created should appear at the bottom of the master channel’s mixer stripe in the Master Mixer window. The default setting for the aux return slider is –Inf; you’ll need to turn the return slider up to hear the aux channel.
The track’s send volume is set by default at 0 dB; you can adjust the send level to control the amount of the track’s signal that is being sent to the aux channel. If, for example, the aux channel contains a reverb effect, the send slider will effectively control the amount of reverb that is applied to the track
The signal sent to aux channels can be taken from three different points in a track’s or group channel’s signal path. Each send can be pre-inserts, pre-fader, or post-fader. The type of each send can be set with the send mode button just above the send pan control in the track’s mixer stripe.
Pre-fader send: when a send is pre-fader, the signal sent
to the aux channel is not influenced by the track's fader settings (volume and
pan). Pre-fader sends may be useful when using an aux channel as a sub-mix,
i.e. to group tracks and control their volume and pan using the aux's return
setting. If a send is pre-fader and the track's volume is set to –Inf, the
track's volume will be set exclusively by the send and return controls.
Pre-inserts send: when a send is pre-insert, the track’s
signal is sent to the aux channel prior to it being processed by the track’s
insert effects and before the track’s volume and pan settings are applied.
Post-fader send: when a send is post-fader, the track’s
signal is sent to the aux channel after it has been processed by the track’s
insert effects and after the track’s volume and pan settings have been applied.
The signal returning from an aux channel to a master channel can be inserted into the mix in four different ways:
Pre-master channel effects
& pre master volume: the signal returned from the aux channel is processed
with the master channel effects and the master volume setting is used
to control the return’s level.
Post-master channel effects
& post master volume: the
signal returned from the aux channel is not processed with the master
channel effects and the master volume setting is not used to
control the return level.
Pre-master channel effects
& post master volume: the signal returned from the aux channel is processed
with the master channel effects but the master volume setting is not used
to control the return’s level.
Post-master channel effects
& pre master volume: the
signal returned from the aux channel is not processed with the
master channel effects but the master volume setting is used to
control the return’s level.
You can use from 0 to 32 aux channels. The higher the number of aux
channels, the more system resources will be used. It's advisable to use only the
minimum number of aux channels that you need.
It’s often useful to adjust the settings of a group of tracks at the same time. Having to change the same setting for many tracks may be tedious; instead, send a group of tracks to a dedicated group channel. Create a group channel by selecting Group/Add Group Channel in the Output to drop-down list in the track’s Properties dialog box. When the group channel is created, a new channel strip appears on the mixer window. More than one track can be sent to a group channel, so that when you adjust the setting of the group channel, all of the group’s tracks are influenced.
A group channel can have its own effects, can
send its signal to aux channels and can have automated volume and pan just like
a regular track. Group channels can in turn be sent to other group channels,
allowing you to organize the song in a hierarchy of groups and allowing for
great flexibility in the routing of signals.
Effects can be automated using either:
· Automation of effects parameters using parameter envelopes or mouse input recording. All VST and AU plug-ins support automation of their parameters, but only some DirectX plug-ins do. All n-Track built-in plug-ins support parameters automation.
or
Click on the volume button
on the toolbar and choose the desired parameter (track volume, pan,
send or return volume etc.). The timeline window will show a line superimposed on
each track.
This line represents the temporal evolution of the selected parameter. When, for example, you select “Draw track volume” from the drop-down menu, you’ll be able to program the track’s volume temporal evolution simply drawing on the track’s timeline representation.
To switch between volume, pan send and return envelopes, click the down arrow next to the volume icon on the toolbar, then choose the desired parameter in the pop-up menu.
To draw the evolution of an effect’s parameter, select “Effects parameters” in
the pop-up menu, then select the desired effect in the left-hand drop-down list
and the parameter in the right-hand drop-down list.
An alternative way to program the evolution
of a parameter is to use fader automation. During playback, click on the
button on the main transport toolbar. When this button is
activated, every action on a mixer’s fader (including volume sliders or pan
knobs, but excluding the master volume knob) will be recorded in the evolution
graph of the relative parameter. If the
button is pressed during playback, the faders will move according
to the relative parameter’s programmed evolution. Once an evolution has been
recorded, it can be edited on the timeline window; display the relative
parameter’s evolution by clicking on the
button on the toolbar.
Holding down the Shift key allows you to freely draw the evolution of the volume (i.e. every move of the mouse will create a new node in the piecewise linear waveform).
Unlike with all the other parameters, the aux returns evolutions are shown for every master channel (i.e. output soundcard) and aux channel in the same screen: even if these evolutions are drawn on top of certain tracks, there's no direct connection between these tracks and the return evolutions, which applies to a particular aux channel and not to single tracks.
If you want to perform a fade in/out, you can select the time interval during which the fade should take place by dragging with the mouse on the track and then selecting the Edit/Volume/Pan drawing/Fade In/out menu option.
By clicking on the
icon on the toolbar, the current selection will be muted; i.e. the
volume evolution will be put to zero. Clicking on the same button while holding
down the Ctrl key (Cmd on a Mac) will destructively silence the
selection: the audio file will actually be modified (you can use the
undo button to reverse this).
Effects parameters automation works with DirectX, VST and AU plug-ins, but while any VST and AU effects will allow its parameters to be automated, not all DirectX plug-ins allow it. To see if a DirectX plug-in supports automation, select “Effects parameters” from the drop-down menu, then select the effect in one of the “Effect” drop-down boxes. The plug-in’s parameters will be listed in the drop-down box on the right. If no parameter is listed the plug-in, doesn’t supports parameter automation.
Sometimes it is useful to apply certain effects only to specific parts of a track or to vary the amount of an effect during the evolution of the track. For example, you may want to increase the amount of a reverb during the chorus of a vocal track while keeping the reverb amount low during the verses.
This kind of processing can be obtained using aux channels and send/return automation.
In the
preceding example, the reverb could be placed in the first aux channel, setting
both the send and the return level for this aux channel to 0 dB. The amount of
the effect can now be regulated by the send envelope. Click on the downward
arrow next to the volume
icon on the toolbar and select "Draw
send to aux 1 volume." Draw the send envelope so that, during the parts
in which you
want the effect amount to be greater, the envelope line is higher.
In a similar way, different effects can be applied to different parts of the same track. If, in the preceding example, you wanted to substitute the reverb effect with a delay only during the chorus, you could put the reverb in the first aux channel and the delay in the second channel. Now you could draw the send envelope so that during the chorus, the send to the first aux goes to 0 (-Inf) and the send to the second aux channels goes from 0 to a suitable level, with the opposite happening after the end of the chorus. n-Track can handle up to 32 aux channels, so sophisticated real-time processing can easily be configured.
Effects, volume and pan envelopes can be
applied destructively to a track’s audio files. Normally, effects are calculated by the
program in real-time during the song’s playback. In many cases, however, it may
be better to apply effects in a permanent manner. For example, sometimes using
too many effects
may cause the computer to run out of resources: applying the effects
destructively can free up CP. In other cases, it may be necessary to apply
different effects to certain parts of a track (this can also be accomplished in
real-time using aux channels send/return automation).
In some situations, it may also be useful to apply volume or pan changes to a audio file in a permanent manner. This typically is needed when mastering a audio file resulting from the mixdown of a song (for example, to apply the final fade out).
To apply destructive effects or envelope processing to a track, select Edit/Apply track effects/envelopes. Select the desired options from the box that pops up:
· Apply:
o Volume envelope: apply the track’s volume envelope to the selected audio file
o Pan envelope: apply the track’s pan envelope to the selected audio file
o Effects: apply the track’s effects to the selected audio file
· Apply to:
o Selection: process the selected portion of the audio file
o Whole file : process the whole audio file
· Convert to:
o Stereo: convert the (mono) audio file to stereo
o 32-bit: convert the file to 32-bit format
· More Options:
o Dither: Use dithering.
o Dither depth: Depth in bits of
the dithering noise.
o Noise shaping: Use noise shaping.
n-Track’s track EQ lets you use up to 20 bands of parametric EQ. Each audio track has 3 EQ knobs on the mixer window. The knobs control the 3 EQ bands’ boost/cut amounts. The three bands are, by default, set as low-shelving, band boost/cut and high-shelving filters, respectively, but each band filter type, frequency and bandwidth can be customized in the EQ properties dialog box.
Even though the
mixer has only three knobs, more EQ bands can be added or deleted in the EQ
properties window using the “add band” and “delete band” buttons. Click the
button on a track to open its EQ
properties window.
The EQ window also has a built-in spectrum analyzer and automatic instrument tuner that can be activated or deactivated by right-clicking on the frequency response graph. Click the “Show/Hide all EQ Controls” button in the EQ properties window to reveal the EQ parameters:
· Boost/cut: Sets the selected band’s boost (when the value in dB is positive) or cut (negative). This function just like the knob that appears on the mixer window.
· Frequency: Sets the frequency at which the band is positioned. The value represents the center frequency for the ‘Band boost/cut’ filter and the corner frequency for the other types of EQ.
· Bandwidth: Size of the band to be boosted or cut (applies only to the ‘Band boost/cut’ band type). The parameter units are ‘octaves’. A bandwidth of 1 octave means that the highest frequency is twice the lowest frequency. A bandwidth of N octaves means that the highest frequency in the band is 2N times the lowest frequency.
· EQ type: Sets the band filter type. By default, the 1st band is a low-shelving filter (which boosts or cuts the frequencies below the corner frequency by the selected amount), the 2nd band a ‘band boost/cut’ filter (boosts or cuts by the selected amount the range of frequencies centered around the center frequency at the width set by the bandwidth parameter) and the 3rd band is a high-shelving filter (boosts or cuts the frequencies above the corner frequency by the selected amount). Other available filters are ‘high pass’ (which entirely cuts all the frequencies below the corner frequency) and ‘low pass’ (which entirely cuts all the frequencies above the corner frequencies).
· Track: The track whose EQ’s parameters are displayed in the dialog box.
· Band: The band whose parameters are displayed in the dialog box.
The spectrum analyzer examines the signal at the output of the EQ and calculates the power of each frequency contained in the signal’s spectrum. The results of the analysis are drawn in the graph. The analysis of the signal is performed using the FFT method. The size of the FFT window, the size of the FFT and number of samples after which each new analysis is performed can be adjusted in the pop-up menu that appears when you right-click on the EQ. You don’t really need to understand what these parameters mean to use the spectrum analyzer; however, here’s a quick explanation:
·
Window size: the window size is the
length of the block of signal that is analyzed. The longer the window, the more
accurate the analysis. More accuracy means that the signal’s frequencies will
be shown with more detail; groups of frequencies close to each other will be
shown as a single frequency band when a short window is used, but will be shown
as distinct bands as the window length is increased. The disadvantage of using
a long window is that the spectrum will show all the signals in the analyzed
block, and if the block is too big, you may get two subsequent signals that you
wanted to be shown separately into a single spectrum.
If, for example, you want to see the spectrum of a sequence of notes played on a guitar, and you set the window length to a number of samples that approximately equals 1 second, if the guitar player plays 3 notes during 1 second, the spectrum of the corresponding signal block will show the frequency bands corresponding to all the three notes that have been played on that second. If some bands overlap, one note may be or contain a harmonic component (an integer multiple) of another, and you’ll only see the sum of the three notes’ spectrums. If a ~0.3 second window is used instead, you’ll see three spectrums, each showing the spectrum of a single note played on the guitar.
A longer window also means that the screen is updated less frequently. This can be overcome by setting the ‘computer FFT every X samples’ parameter to a size less than the window’s. This, in turn, means more analysis per second, and thus the CPU will be used more intensely.
· FFT size: the size of the FFT must be at least the size of the window. A longer size for the FFT basically implies a smoother graphical representation of the spectrum in the graph, but the accuracy of resolving adjacent frequency bands will not improve unless the window size is also increased.
· Compute every # samples: if the window size is large, it may be useful to set this parameter to a size smaller than the window size. When this is done, the program will analyze the signal at a frequency higher than that set by the window size (which implies that a same portion of the signal will be analyzed more than once). More analysis per unit of time imposes a greater stress on the CPU.
· Window type: without getting too technical, the choice of the type of the window used is a tradeoff between the smoothness of the spectrum and the ability to resolve (show as separate) small peaks in the frequency spectrum. The rectangular window is the best for the latter requirement, while the Blackman window provides the smoothest spectrum.
You can enable or disable the spectrum analyzer by right-clicking on the frequency response window. With stereo signals, you can display the spectrum of the left channel, the right channel or both channels simultaneously (in the last case, two spectrums are drawn, the left channel’s in red and the right channel’s in green).
You can enable or disable the tuner by right-clicking on the frequency response window. The tuner analyzes the signal at the output of the EQ and shows how close the signal’s main frequency component is to the frequency of a musical note. The closest note is automatically selected so that, for example, when tuning a guitar, you don’t have to manually tell the tuner which string is being tuned.
When the note is close but not exactly equal to the note’s frequency, the percentage of the difference in frequency will be shown, allowing you to adjust the instrument (typically a guitar) to tune it to the desired note.
The accuracy of the tuner is typically so high that it’s likely that you won’t be able to reliably tune the instrument to the exact note frequency. With a guitar, the tuner detects even small detuning due to the bending of the guitar’s neck caused by the weight of the hand resting on top of it. This doesn’t mean that the instrument is not in tune, as the human hear won’t typically notice a very small detuning, and a musical instrument is usually built to be heard by a human ear and not by an electronic tuner!
The View/Signal Path menu command opens the signal path window, which shows the flow of the audio signal inside the audio engine.
The signal path view is interactive. You can:
· drag a track’s output from one output to another
· bypass sends
· bypass effects
·

open tracks or plug-ins context menus
by right-clicking on the plug-in box
Mastering is the last step in the production of a song. Once the song has been mixed down to a single audio file, the file may be loaded again to apply the master processing (typically EQ and Compression) to produce the final audio file to be used for burning a CD.
When mixing down a song, if you plan to process the resulting audio file (which is often advisable), you may want to do the mixdown with 32-bits resolution, so as to assure the minimum degradation of the sound quality due to the repeated processing (even when using 16-bit audio files, the sound quality degradation caused by a few processing steps is very hard to notice. When using 32-bit audio files, you may feel free to process the same file tens of times without any noticeable loss of quality).
n-Track Studio can use dither and noise shaping when converting signals from the program’s internal format (32-bit) into 16- or 24-bit format. This conversion takes place at the very end of the signal chain, when the signal is prepared for the output to a soundcard or for writing to a WAV file, for example during the mixdown.
The dithering process consists of adding a very small and calibrated amount of noise to the signal before converting it to a less accurate representation. This can increase the dynamic range of the resulting signal, allowing it to represent signals even smaller than a given format’s theoretical maximum resolution. This process also reduces the effects of the non-linear distortion inherent to the quantization process, which can result in the generation of harmonics that are often annoying to the human ear.
The drawback of the dithering process is that a small quantity of noise is added to the original signal. The amount of noise added is controlled by the Preferences/Options/Dither Depth parameter. This parameter is expressed in bits, which are referred to the current output format. For example when converting to 16 bits, 1 bit unit is 1/(2^16) of the full scale, while when converting to 24-bits, 1 bit is 1/(2^24) of the full scale.
The noise introduced by dithering can be reduced using the noise shaping technique, which moves the dither noise to a region of the frequency spectrum where the human hear is less sensitive (the high frequencies).
The noise shaping process should typically be applied only when preparing a WAV file to be used for creating a CD track, when no other processing is to be applied to the file. The benefits of the noise shaping process are in fact somewhat delicate, and can be completely voided by additional processing.
n-Track Studio uses numerous types of files for different purposes.
Song file
Extension: .sng
Song files are n-Track Studio project files. They contain the song structure, the mixer settings, the effects used on each track, the volume envelopes, and so on. Song files don’t contain the actual audio data of the audio tracks: they instead refer to WAV or Aiff files that contain the audio data. To play back a song, you’ll need both the .sng file and the associated .wav files.
WAV and AIFF files
Extension: .wav, .w64, .aif, .aiff
WAV or AIFF files (commonly referred to as ‘audio files’ or ‘wave files’ in this guide) contain the audio data that makes up tracks’ audio signals. Each waveform that appears on the timeline window corresponds to an audio file (the full path of the file can be seen in the audio file’s properties dialog box that appears when you double-click on the waveform). A song can contain more than one instance of the same audio file. Audio files can have many formats. n-Track Studio supports stereo and mono audio files of any sampling frequency, with 16, 24 or 32-bit depth. n-Track also supports Wave64 files which allow for file sizes greater than 2 Gb.
Standard MIDI file
Extension: .mid
Standard MIDI files contain sequences of musical notes. The main difference from .wav files is that while .wav files contain a representation of an acoustic signal, MIDI files contain the sequence of the notes played on an instrument. For example, a MIDI track of a piano recording contains only the sequence of notes (described as note number, velocity with which the key was pressed, time at which it was pressed and duration of the note) played by the pianist. The corresponding .wav file contains the acoustic signal emitted by the piano. The difference between a .wav and a MIDI file is basically the same as between a recording of a concert and the concert’s sheet music. The .wav file obviously contains much more information than just the sequence of notes; for this reason, .wav files are typically much bigger than MIDI files. n-Track Studio can both import standard MID files into MIDI tracks and export the current song’s MIDI tracks to standard MIDI files.
Peak file
Extension: .npk
n-Track Studio creates peak files to speed up the displaying of audio files’ waveforms on the timeline. Normally, the program generates the .npk files during recording. These files are not critical, as the information in them is contained in the audio files they were created from. If you accidentally delete an .npk file, the program will recreate it the next time it has to display the associated audio file’s waveform.
Packed song file
Extension: .sgw
Packed song files are a special kind of song file. Like an .sng file, an .sgw file contains the structure of the song. A packed song file, however, also contains the audio data used for the song’s tracks. Thus, a .sgw file can be used to save and transfer a whole song as a single file, without having to worry about transferring the .wav files associated with the .sng file. Packed song files are mainly useful for archiving or transferring songs from one computer to another. When the program opens a .sgw file it still has to recreate the .sng and .wav files structure, so the .sgw format is not useful for saving songs while you’re actively working on them. The audio data contained in .sgw files can be either uncompressed or compressed.
MP3 compressed audio file
Extension: .MP3
MP3 audio files are compressed to drastically reduce the size of the file, while only marginally sacrificing the audio quality of the recording. MP3 files are typically a tenth or less of the size of uncompressed .wav files, and are widely used to transfer songs over the Internet. n-Track Studio can save a song’s mixdown to MP3 or convert existing .wav files to MP3.
Ogg Vorbis file
Extension: .ogg
Ogg Vorbis compressed audio files. See the Ogg Vorbis files topic.
Windows Media Audio file
Extension: .WMA
Windows Media Audio is a compressed audio file format similar to MP3, but WMA files are typically ˝ or less of the size of MP3 files of comparable quality, and are thus more suited for transferring songs over the Internet. Unfortunately, .WMA is a Microsoft proprietary standard, is available mostly on Windows PCs, and is not currently nearly as popular as the MP3 file format.
Tracks can be either mono or stereo. On a stereo track, the pan slider will simply act as a balance control, while it will actually pan a mono track to the left or to the right.
You can set the playback number of channels (stereo or mono) in the dialog box that appears when you click on the playback VU meter’s settings button. Listening to a song in mono mode is often useful to ensure that the song is mono-compatible, i.e. that it will not sound bad when played on mono devices (old radios, mono televisions, etc.).
It's also possible to disable stereo playback only while recording other tracks: in fact, many soundcards are capable of full-duplex (simultaneous recording and playback) operation only when used in mono mode. If you get messages such as "Error opening soundcard” when trying to start the recording, enable the "mono playback while recording" option in the Preferences/Recording settings dialog box.
n-Track Studio normally saves projects into .sng files. An .sng file contains only the structure of the song, not the audio data, which is kept in separate .wav files (see File Types).
Saving a project as a packed song (using the .sgw extension) allows you to save all of the song data, including the audio data, into a single big file. This can be useful for transferring songs by email or for archiving them to CDs.
The .sgw format is not suited for everyday saving of projects, as n-Track will always need the .sng and .wav files before it will allow you to work (i.e. edit, playback and record) on the song. Each time you load an .sgw, the program will extract the tracks’ audio files and will let you save the song’s .sng file.
The audio data stored in an .sgw file can be either uncompressed (producing a perfect copy of the song) or compressed using Ogg Vorbis compression. When compressing, the file size drops significantly at the cost of losing some information on the audio data, with a consequent slight decay in sound quality (the amount of which can be selected using the compression slider in the Save .sgw file dialog box).
Once you have finished adding tracks and
you have adjusted the volume and pan settings, you’ll want to mix down all the tracks to a single wav file. Select
"Mixdown song" from the File menu and choose the desired
options in the Box. This operation may take a while.
You can also select “Mixdown while playing,” and the program, after you enter the name of the file to mixdown to, will start the playback. You will hear the song played normally, but the playback data will also be saved to the mixed down file. In this way, you can adjust volume and pan settings, as well as add or remove effects, while the song is playing. The resulting mixed down file will exactly reflect what you hear during this “final mix down.” I it doesn't matter if the computer isn't fast enough and every now and then you hear some jumps in the music: the mixed down file will not contain these defects, as these jumps are caused by the computer not being able to furnish data to the soundcard in time for its playback. When it finally manages to have this data ready, it saves it to the mixed down file just after the previous good packet of data.
Mixing down works only on audio tracks, so if you have included a MIDI file, you'll have to "record" it by selecting MIDI as the recording input in your soundcard mixer, so the program will be able to treat it as a WAV file.
On slow machines, or when the number of tracks you’re using becomes high, it may be a good idea to mix down some or all of the tracks to one WAV file, and then proceed to record other tracks. In this way, you’ll be able to manage more tracks, and you won’t suffer from possible synchronization failures that may happen when the computer is too loaded by the high number of tracks.
Once the mixing down has finished, you may want to convert the WAV file to MP3 or WMA format for distributing the song in a smaller file.
You can use n-Track to mix songs in a surround format or to create a surround-sound soundtrack to a video. n-Track supports all popular surround formats, including 5.1, 6.1 and 7.1. The majority of film and audio DVDs use the 5.1 format.
To create a surround project:
· Select the Settings/Audio devices menu. Make sure that the desired physical output channels are selected in the box that appears. If you select a multichannel WDM, WaveRT or Asio audio output driver, select the Settings/Soundcard settings/Playback format menu command, click on Select I/O channels and make sure that all the desired outputs are active
· Select an audio track. Right-click on the audio track and select Surround from the pop-up menu. The track will automatically be sent to the Surround Main output.
· Right-click on the circular surround pan control that has been added to the track mixer strip. Click on the surround format name (for example “Surround 5.1”) in the upper right corner of the surround pan window. The surround configuration dialog box will appear, allowing you to select the desired surround configuration (5.1, 7.1, etc.), which surround channels go to which physical audio outputs and the order of the various surround channels (the central channel is usually the 4th channel, but since your setup might be different, you can use the up/down buttons to change the relative ordering of the channels). You can modify the surround configuration even after you have started working on the project.
The Surround Main output is a virtual channel that maps to the actual physical output channels according to the current surround configuration.
Surround effects plug-ins can be applied on Surround Group channels: select a surround track, right-click on it and select “Send to a new group” from the pop-up menu. A new group channel will be added to the mixer.
Effects applied to the surround group channel will need to be able to process surround signals (i.e. regular mono or stereo effects will not work or will not produce the expected results).
When the project is complete, you can create a single mixed down audio file:
· Select the Mixdown command from the File menu
· Click on the More options button
· Select Single multichannel audio file from the output format drop-down list in the lower right corner of the window
The most common surround format is 5.1. Increasingly common formats are 6.1 and 7.1. The 5.1 format is the standard multichannel audio format for movie DVDs. Audio on video DVDs are usually encoded in either Dolby Digital or DTS compressed formats. Multichannel compressed formats allow you to store 6 channels of audio data in less space than 2 channels of uncompressed audio data, such as the uncompressed format used in audio CDs.
The output of a surround project created with n-Track will be in uncompressed PCM WAV format. The mixdown audio file will need to be imported into a DVD-authoring program that will use it to create the audio track of a DVD by encoding it to a multichannel compressed format.
The basic version of both Dolby Digital and DTS allow up to 5.1 channels. Extended versions of both standards, named Dolby Digital EX and DTS-ES, allow the encoding of 6.1 channels.
The “.1” in a surround format specification stands for the LFE channel. LFE is an acronym for “Low Frequency Effects.” In contrast to the main channels, the LFE channel contains only low frequencies below 120 Hz. The LFE channel is used to drive the sub-woofer, which is used in all home-theater speaker systems.
One of the main reasons why the LFE channel has been added to Dolby Digital and other multichannel digital standards is for compatibility with legacy analog film audio formats. LFE channels were used in analog film audio for technical (unburdening the main channel tracks from the distortion easily caused by heavy bass signals) and economical reasons (eliminating the need to install expensive analog cross-over filters to separate the signal going to the subwoofer and to the main speakers) that don’t matter in the digital world.
Some confusion usually arises between the concepts of LFE channel and subwoofer. The subwoofer is typically used for the reproduction of both the low frequency content of the signal of the main channels (left, central, right, surround left, etc.) and for the reproduction of the LFE channel. On many small home theater setups, the speakers can’t reproduce low frequencies, so the home-theater amplifier divides the signal of each of the main channels, sending the high frequency content to the corresponding main channel speaker and the low frequency content to the sub-woofer.
It follows that you don’t typically have to do anything to have a signal go to the subwoofer. You can totally ignore the LFE channel and the sub-woofer will still play powerful bass sounds according to the bass content of the main channels. The LFE channel allows for extra control and separation of bass sound effects. LFE content is typically present in movie soundtracks only during scenes containing sounds such as explosions, airplane engine noise, etc.
It’s generally believed that human ears can’t detect the spatial origin of a bass sound. That’s the reason why, in home-theater speaker systems, the sub-woofer’s position in the room is considered irrelevant. Accordingly, the n-Track surround panner doesn’t place the LFE channel as a speaker in the virtual room shown in the panner screen. The LFE channel has a separate slider that controls how much of the source signal goes to the LFE channel.
The mixdown created with n-Track can be used to create a DVD that will play on standard DVD players connected to a home theater speaker system.
A very high-quality audio standard (24-bit 192 kHz) exists for audio-only DVDs. Unfortunately, the DVD-Audio standard is not yet widely used, and only a minority of DVD players can play DVD-Audio discs. An alternative way to create audio-only DVDs is to create regular video DVDs with no video (or just a static background image). Although the audio quality will not be as good as with DVD-Audio, the discs will be playable on any standard DVD player.
Audio on standard video DVDs is typically encoded in Dolby DigitalTM AC3 format. Third-party software that can be used to convert the n-Track mixdown to an AC3-encoded file includes:
· BeSweet
The resulting .ac3 file can be imported into a DVD authoring program to create and burn the actual DVD.
Some DVD authoring or Dolby DigitalTM AC3 encoder programs require a single .wav file as input for each of the surround outputs. You can create single WAV files for each output by selecting the Two mono WAV files for each output option in the mixdown dialog box.
The n-Track mixing algorithm is conceptually very simple: the tracks are first fed through the
inserts effects and then mixed together, each one amplified by a factor
resulting from the combination of the master volume, the track volume slider
and the track envelopes. The volume evolution is used at the very ending of the
mixing process so it doesn’t influence the aux sends, even if the “pre-fader
sends” option is unchecked. After the tracks are mixed, everything goes to the
master channel effects. Sends and returns are placed at different points
depending on the Aux dialog settings.
64-bit processors are widely available and today constitute the majority of CPUs on newly sold computers. 64-bit operating systems are available (starting with the x64 version of Windows XP).
Windows
Despite the great availability of 64-bit CPUs, 64-bit operating systems still not quite popular, and the vast majority of users still use the 32-bit (x86) version of Windows XP, 7 or Vista.
n-Track for Windows is available in two “formats”:
The 32-bit (x86) version runs on both the 32 and 64-bit versions of Windows.
The 64-bit (x64) version only runs on the 64-bit versions of Windows. The standard version of Windows XP is not a 64-bit operating system and can’t install or run the 64-bit version of n-Track.
Mac
Mac OS X versions have included 64-bit support since 10.5 (Leopard); however, the 64-bit version of n-Track for Mac only supports 10.6 (Snow Leopard) and later. With Mac OS X, there is a single version of the operating system that can run both 32 and 64-bit programs.
The 64-bit version of n-Track can only use 64-bit native plug-ins. If you need to use legacy 32-bit plug-ins, please use the 32-bit version of n-Track.
If you’re unsure which version of n-Track to use, go with the 32-bit version, as it is the most compatible version and has no noticeable performance disadvantages over the 64-bit version, unless your songs use a lot of instrument plug-ins that require massive quantities of memory.
Both the 32 and 64 bit versions of n-Track are installed on your Mac Applications folder, they have the same icon, the 32 bit version name is just n-Track, the 64 bit version is named n-Track64.
24-bit recording and playback is currently available only on very good quality soundcards, such as the Creative Audigy 2 and 4, M-Audio Delta cards, Echo Layla, EMU 1820, etc. If you don't have such a card, you don't need the 24-bit n-Track version. If you'll buy one of these cards later, you'll be able to upgrade to the 24-bit version for the exact price difference between the standard and 24-bit version.
If you don't know if your soundcard can handle 24-bit recording and playback, it probably can't. The Creative Labs Soundblaster 16, 32, AWE32/64, Ensoniq/AudioPCI 64, SB128, Live!, Audigy 1, and most of the multimedia soundcards that you find pre-installed on your computer are only capable of 16-bit recording and playback.
The Soundblaster Audigy 2/4 can record and playback in 24-bit (and 96 kHz) mode.
The Soundblaster Audigy 1 can only record and playback in 16-bit mode, despite the specifications that may lead you to think that it's a 24-bit card.
To check if a soundcard accepts the 24-bit format, click on the settings button on the playback VU meter window, select "24-bit", click OK and start the playback. If the format is not supported, the program will report an error. Make sure you've selected the soundcard's WDM or Asio (not MME) drivers in the File/Settings/Preferences/Audio devices dialog box.
The format of the input or output (16- or 24-bit) has no influence on the precision with which the program does its internal signal processing calculations. The program always uses 32-bit (or 64-bit if the option is selected) floating point signals for optimum sound quality and dynamic range.
There are a number of settings you can customize to make the program work better with your computer. You can alter some parameters to make the program more or less “heavy” on the computer CPU, so that if you have a slower CPU you can use the program without problems.
Experiment with different playback buffering settings to find the optimum compromise between reliability and “real-timeness” of the program’s operation: fewer buffers of a smaller size will allow the result of actions made during playback, such as altering the volume, pan, or effects parameters, to be audible in less time. Low buffering can, on the other hand, lead to occasional jumps in the playback and loss of sync which can be very disturbing if they happen during recording. Consider keeping the buffering heavy during recording, then, during the final mixing phase, when real-time response is more important, using smaller buffers.
On slower computers, or when the number of tracks becomes great and the system becomes heavily loaded, you can reduce the load on the processor:
· Disable the mixer VU meters (uncheck the Preferences/Appearance/”Show VU meters for each track” option)
· Disable automated volume/pan evolution for MIDI and audio tracks
· Turn off or resize to a small dimension the recording and playback VU meters
· Disable the “Expand mono tracks to stereo” option in each track’s properties dialog box
· Use higher program priority (use caution with very high priorities)
· Disable the display of waveforms during recording
One strategy for working with a great number of tracks is to mix down a partial version of the song.
Suppose your system can’t seem to handle more than 8 tracks: you can record the first 8 tracks, adjust the settings and mix them down to a single audio file without deleting the original audio files. Then you record the next block of tracks using the partially mixed down file as the base. Once all tracks have been recorded, you can reload all the audio files that were previously mixed down, adjust all the settings and mix down all the original tracks into the final song.
The colors and fonts used by the program can be configured through the Preferences/Appearance dialog box.
The background of the mixer window, of the main window and of the timeline window can be customized putting a bitmap file in the program’s directory, named respectively “mixer_background.bmp”, “main_background.bmp”, and “timeline.bmp”. If, upon loading, the program finds one of these files, it will use it, otherwise it will use the standard background.
The Toolbars can be customized by double-clicking on them or via the Preferences/Appearance/Customize toolbar 1/2 buttons: buttons can be moved, added, deleted and moved from one toolbar to the other. Toolbar buttons can also be moved in this way: hold down the Shift key, click on a button and drag it to the desired location (within the same toolbar). Releasing the button outside the toolbar will delete it.
To record a new MIDI track, click on the
microphone
button on the toolbar. This button
controls which source (audio, MIDI and audio/MIDI) the program records from.
After clicking on it, the button will change its image to a keyboard
(which indicates MIDI recording). Pressing
the button again will change the image to a keyboard plus a mic
, which means that both audio and MIDI
tracks will be recorded.
Make sure that the MIDI instrument you want to record from is connected to your MIDI input device (typically a soundcard’s joystick/MIDI connector), and that the MIDI input device is selected in the Settings/Preferences/MIDI/MIDI devices dialog box.
If you want to hear the notes played with the MIDI instrument through the current MIDI output device, the program can route the incoming MIDI signals to a MIDI output device. To enable this, check the option for Always open and select either the Auto or Manual MIDI echo mode. See the Preferences/MIDI/MIDI devices topic for info on how the MIDI echo modes work.
Sometimes it is useful to record more than one MIDI track at a time, as when transferring MIDI tracks from a MIDI keyboard. You can record multiple MIDI tracks at once using the following procedure: create an empty track for each track that you want to record (with the Add Channel/Add new blank track/MIDI menu command), then, for each track, change the “all channels/all devices” setting in the MIDI track's properties dialog box to the channel from which you want to record the track.
Editing MIDI
tracks can be useful for correcting imperfections
in recorded tracks or for creating new tracks from scratch. MIDI tracks can be
edited in the Piano Roll window: select a MIDI track and press the
button on the toolbar (if no MIDI
track is selected, a new MIDI track will be created).
The Piano Roll view can also be activated by clicking the Piano Roll button on the track properties dialog box which appears when you double-click on a MIDI track.
The Piano Roll
window shows the MIDI events occurring in the track. The most common events are
notes, but other types of events (controller commands, pitch bends, etc.) can
be displayed and edited in the same way as notes.
![]() |
You can set
which types of events are displayed and which events are inserted in the dialog
box that appears clicking on the events display
icon on the piano-roll toolbar. The Place
drop-down box lets you select which event will be inserted when you click on
an empty spot in the piano-roll window. The View box lets you choose which events will be displayed.

The event properties can be manually edited in the dialog box that appears when you double-click on an event rectangle.

Multiple events can be selected and
moved at the same time. New events can be placed activating the
icon on the toolbar (click while holding
the Windows Ctrl or Mac Cmd key to set the properties of new notes), while
events can be deleted when the
button is activated.
Cut, copy & paste operations can be performed on MIDI events
by selecting the desired events or by selecting the desired temporal interval
(dragging with the mouse on the time axis). Once you have selected a group of
events, click on the
copy button on the toolbar, select the
offset where you want the notes to be copied and press the
paste button to paste the notes into the
new position. Cut, copy &
paste operations can be especially useful when composing MIDI drum tracks.
Create a new blank MIDI track with the Track/MIDI/Create blank MIDI track menu command. Assign the new track to channel 10 in the track’s properties dialog box. This is very important because channel 10 is the channel for drums.

Open the piano roll. The left side of the piano roll window usually shows a vertical piano keyboard. This is often useful when composing or editing tracks of melodic instruments. Drum instruments are different from the usual MIDI instruments, however, because each “note” of a Drum channel corresponds to a different percussive instrument (for example D4 is a high tom, D3 an acoustic snare, etc.) When editing a drum part, it is often useful to have the names of each drum instrument appear in the piano roll window.
Right-clicking on the left part of the piano roll window opens a dialog box in which you can select how the notes are drawn. If “Show piano” is selected, a vertical piano will be drawn.
If any other entry is selected, the program will write each note name as set in the MIDI instrument definition. Instrument definitions can be edited in the dialog box that opens clicking on the 'Instruments' button in a MIDI track's properties dialog box, then on 'Edit'. The name of a note can be modified by double-clicking on its name while holding the Ctrl or Mac Cmd key. For more info, see MIDI Instruments assignment/definition.
Select the note button on the toolbar:![]()
Clicking on the piano roll, create a short sequence of beats that makes up the base rhythm that you want to repeat over the length of the whole song.
Click the arrow button on the toolbar: ![]()
Select the desired area you want to copy. Once you have finished your selection, simply click the copy button. If you prefer, you can use the keyboard shortcut Ctrl+C (on a Mac, Cmd+C).

Now choose the offset to which you want to copy your pattern
selection and press the paste button. As you can see, your selection has been
copied to the offset you wanted. This procedure is particularly useful if you
want to create drum tracks in which you must often manage repeating measures.

Instead of manually pasting the same beat multiple times, you can let the program automatically replicate the selection you’re pasting by holding the Shift key when you click on the “Paste” button or by pressing Ctrl+Shift+V (Cmd+Shift+V on a Mac).

The mixer faders and the transport controls (play, record, etc.) can be controlled by a hardware MIDI control surface. You can configure what actions are associated with the MIDI events generated by the control surface faders and buttons in the Settings/MIDI faders – control setup dialog box.
The dialog box shows a list of the defined events and the associated actions. To change an event, click on its name in the list, change the desired options in the lower part of the window, then click on an empty part of the list window to save the settings.
If you don’t know which events are generated by your control surface, you can select the event you want to edit, click on the “Learn” button, then move the control on the control surface. The program will detect which event the control sends and update the “MIDI event” settings accordingly.
Use the options in the dialog box to set up the faders on your MIDI device:
· MIDI event: The type of MIDI event with which to associate the desired action
· Channel: The channel to which the MIDI event is sent
· Controller/Note: Number of the MIDI controller (if the event is a MIDI controller) or of the MIDI note (if the event is a note-on or note-off event).
· Learn: Automatically detect the event generated by the control
· Range: Specifies the range of values affected by the control
· Action: Action (change in volume, pan, mute, solo etc.) that you want to associate with the selected MIDI event
· Track: Track to which you want to apply the selected action, or output channel number (if the action is return volume or return pan setting)
· Aux: Aux channel to which to apply the action (used for send vol, send pan, return vol & return pan actions)
· Receive/Send as: Check the Receive option if you want the program to respond to the MIDI events coming from the control surface. Check the Send option if you want the program to send MIDI events to move the control surface faders when the faders are moved within the program (useful with motorized control surfaces).
VST/DX instruments are VST/DX plug-ins that accept MIDI data as input and can be used as MIDI synths. Using a VST/DX instrument MIDI synth has many advantages:
· The synthesized signal is mixed as a regular audio track, so it can processed with effects, sent to aux channels, etc.
· The synchronization is (unlike that of MIDI devices) sample-accurate, so the audio and MIDI tracks will always be perfectly synced
·
Since all the MIDI output can be sent to one or more VST/DX
instruments, the program doesn’t even need a soundcard with MIDI capabilities.
The quality of the MIDI tracks depends exclusively on the quality of the VST/DX
instrument synths.
How to use a VST/DX instrument:
· Select the desired VST or DX instrument to use from the Add Channel / Add New Instrument Channel menu.
· Open a MIDI track’s Track properties dialog box. The Output port drop down box should now contain, besides the regular MIDI output port, an entry for the VST/DX instrument plug-in.
You can use VST/DX instruments on one track.
An entry for each plug-in will appear in the MIDI track’s properties output
port drop-down box.
Both the instrument plug-in’s audio channel track and the MIDI track’s volumes and pan settings can influence the signal.
Instrument plug-ins can be used to play a MIDI instrument live (i.e. let the VST/DX instrument output the notes you play on a MIDI keyboard in real time). To enable live playing through an instrument plug-in, turn on live input processing mode by clicking the Live button on the toolbar. Set the Echo Mode in the Preferences/MIDI/MIDI devices dialog box to either ‘auto’ or ‘manual’ (if the echo mode is set to manual, you’ll have to select the desired instrument plug-ins in the output drop-down box). Press a note on the MIDI keyboard to hear it play through the synth plug-in.
The delay between the time that a note is pressed on the MIDI keyboard and the time you actually hear the note played is called latency, and is caused by playback buffering. With soundcards with WDM, WaveRT or Asio drivers, the buffering can be usually made small enough so that the latency is not noticeable. Playback buffering can be adjusted in the Buffering settings dialog box.
Each MIDI channel of each MIDI output device can be assigned to a different MIDI instrument. A MIDI instrument is a set of program names, note names and controller name definitions that will be used for the selected MIDI channel/output port combination. The MIDI instruments assignment dialog box can be opened by clicking on the “Instruments” button in the MIDI track’s properties dialog box. The left box lists all 16 MIDI channels for each MIDI output port. When a channel is selected, the relative instrument is selected in the right box.
The MIDI Instruments Definition dialog box lets you define program names, note names and controller names for your MIDI instruments. MIDI instruments can be assigned to MIDI output ports/channels by clicking the ‘Edit’ button in the MIDI Instruments Assignments dialog box.
The MIDI instruments definition dialog box has two boxes. The left-hand box shows the current instrument banks, while the right-hand box shows the currently available program names, note names, and controller names sets.
The entries in the right-hand box are the items of which instruments are made. The items can be dragged from the right box to the left box. If, for example, you want to assign the program name set ‘General MIDI’ to the selected instrument, select ‘General MIDI’ in the right-hand box (it will be listed under the ‘Program names set’ folder), then drag it to the desired position in the left-hand box.