Minimum System Requirements

To run n-Track Studio, you need the following minimum configuration:

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

Installation

Download the installer file from www.ntrack.com.

Once done, double-click the downloaded file to start the installation.

Online Activation

n-Track Studio does not require registration codes. If your computer is connected to the internet, all you need to do to activate your version of n-Track is to click on the Activation command found in the Help menu, and login with your n-Track credentials.

Once you login, you'll see a list of the products that you have purchased and that are ready for activation. To activate a product, simply press the Activate button found next to it.

n-Track studio will now be registered with the name and credentials you provided. Having a registered version of n-Track not only unlocks all the software's features but also gives you other benefits as a registered user.

Offline Activation

To activate n-Track running on a computer that cannot be connected to the internet you can use the Offline activation method.

We strongly recommend, when possible, using the Online activation method instead as the Offline method is more prone to errors. After the online activation, n-Track doesn't require an internet connection, so if you can temporarily connect the computer to the internet, you can use the online activation.

If you cannot connect the computer to the internet, not even for even a short time, to activate offline:

• open the Help menu and click on the 'Activation' command while holding down the SHIFT key.
• a dialog will open. You'll find an alphanumeric code in the first text-box. This is the device id that you must send us in order to receive your offline activation code.
• to receive your activation code, simply login to your n-Track account on the ntrack.com website from any device with an internet connection, and click on 'My purchases'. Copy and paste the device id into the textbox at the bottom of the page, click 'Get activation code' and wait for your activation code to be returned.
• once you receive your offline activation code, copy and paste it to the Offline activation code text box inside the n-Track application.
• press the 'Activate' button, and wait for the response that will be displayed in the text box on the top of the window.

If something goes wrong and you see an error message, copy and paste the message in the content of an email with subject Offline activation Failed, and send the email to support@ntrack.com.

Uninstallation

Deactivating and moving PCs

Depending on which edition of n-Track you have purchased you'll be able to activate the app on a certain number of PCs and/or Macs. If you reach the maximum number of available activations, and want to install n-Track Studio on a new computer, you'll need to manually deactivate one of your copies on one of the PCs where you previously activated n-Track.
To deactivate your copy please launch the app, select Help from the top menu, then select Activation.
Uninstalling the software does not automatically deactivate your version.
If you have already uninstalled n-Track on the PC where you want to deactivate, you can simply reinstall it, deactivate it (from the Help -> Activation menu) and then uninstall n-Track again.

Please remember to always deactivate n-Track on any PC that you dismiss, sell or before reinstalling the operating system.

Finding initial inspiration

The process of making a song can be carried out in many different ways and following very different steps and criteria. You may already have a melody or a chord sequence in mind, or you may want to start from a beat or a simple instrumental loop and then build your song on top of that.

Although n-Track Studio will support whatever creative workflow you may prefer, it also provides some tools to quickly and effortlessly start laying down the foundations for a song. The Loop Browser, accessible by pressing the button on the main toolbar, let’s you add instrumental and drum MIDI and audio loops, directly into your project. In the Loop Browser, selecting MIDI —> Quick Groove lets you add a drum track, a bass track or a bass+drums track directly into your project, ready for you to jam on or start building your song.

When you select the Quick Groove pattern you want to add from the Loop Browser, n-Track will automatically create a Step Sequencer track with the selected pattern already loaded and ready to play. You can choose from n-Track pattern presets or make your own. You can then edit the patterns, re-arrange them or create new ones.

Recording a Demo Song

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 Vista, 7 and later, 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, System Preferences/Sound/Input), then check that the mic in is the only input source selected, and that the level is sufficiently high.

See also:
Setting the Recoring level

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.

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" (be sure that the vu-meter size enable you to see this icon, in case it don't, enlarge the vu-meter)

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 coloured in red. Complete the recording and click on the "Stop" button.

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.

Re-Recording Portions of a Track

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 Track

Now 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. Be sure that the aux channels are showed (click on the "select tracks to view" icon then "Show Aux channels")

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 thefirst 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: y 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).

Adding Compression

Let's assume that all went well and the tracks have been correctly recorded. Now you notice that the vocal track 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 geeral and of n-Track in particular. Effects plug-ins can be added from the effects browser (choose "compressor").

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.

Applying an Effect to Specific Parts of a Recording

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.

Final Mix-Down, Compression, EQ and Reverb

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.

Programming Volume Envelopes

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 . You can now select which envelope to draw by choosing it from the Envelope options panel on the track left panel



or by adding an envelope track by clicking on the icon. 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. Right click the selection and select 'Silence Selection' from the menu 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.

Fading out the Song

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 (right click on the toolbar then "add/remove" to add the fade out button).

Mixing Down to a Single Audio File

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 open a new project. Import the mixed-down audio file using the Track/Import audio file menu command. Listen to the track to hear how your mixdown sounds.

Editing the Mixdown File

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.

Burning to CD

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.

Setting the Recording Levels

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").

Signal Levels & Clipping

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.

Soft Clipping

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.

Audio Devices Selection Dialog Box

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.
Learn more about selecting audio devices in the Audio devices tutorial video (Windows).

Watch Video Tutorial >

Wasapi - Introducted with Windows Vista, Wasapi is the current Windows official standard for low latency audio. Wasapi can theoretically achieve low latency in both shared (i.e. multiple apps accessing the audio device) and exclusive mode. Exclusive mode is however recommended for pro-audio.

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 Vista and later 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 macOS. CoreAudio allows for potentially very low latency audio even when multiple apps are accessing an audio device.

Using WDM, WaveRT, Wasapi, 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).

Selecting a driver in the Audio Devices dialog box will typically enable all of the soundcard’s channels. You can limit the number of channels that you want to use in the Soundcard settings -> 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 Technologies → 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.

Recording More than One Track at a Time

n-Track supports simultaneous recording of multiple audio 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.

Most standard 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.

Audio devices specifically built for audio recording usually have two or more analog inputs. On each one you can connect either a mic, an electric or electro-acoustic guitar or bass or line level instrument (such as a keyboard synth or electric piano). Often each input has a line vs instrument switch.
If your audio device is a generic model, not specifically designed for audio or music recording (such as the audio devices built into the vast majority of laptop and desktop PCs) you can still record two instruments playing together, although you'll need to connect one instrument to the left channel and the other to the right channel of the soundcard’s stereo line input connector. You may need a splitter cable to combine the two mono inputs into a single stereo line input.

Recording Using Take Lanes

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 the track to play a portion of one take and the rest from another take:

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).

Count-in, Punch-In and Multiple Takes Recording

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.

To activate the count-in, click the button on the main transport bar. When count-in is active and press rec (or also play if Count-in on playback is active) the playback starts one or more measures (as set with the Set count-in measures setting) before the playback start offset in the song.
You can also select if there should be a metronome click on the count-in before song playback from [Metronome on count-in].

Each punch-in take is recorded to a new take on the same track. The time at which the start and the end of the recording will occur is determined by the current selection. 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.

To configure the punch-in settings, Shift+Click on the button. The Metronome Settings box will appear, with the punch-in settings in the bottom left, which contain the Count-in and Punch-in options.

If Enable punch-in recording is active, and count-in is disabled, you can start the recording at any position before the selection, the playback will start immediately while the recording will start at the start of the selection and finish at the end of the selection. When punch-in is active n-Track will record a single take only.

Recording multiple takes: if the loop button in the main transport toolbar is active, and a selection is active on the upper time axis, the recording process will automatically restart when the cursor reaches the loop 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. Count-in can be enabled for multiple takes recording, in that case on each loop iteration the playback will start the set number of count-in measures before the selection start. If the Enable punch-in recording option is selected only one take will be recorded.

Voice/Level-Activated Recording

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.

Destructive Audio Editing

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.”

Non-Destructive Audio Editing

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.
See also:
Editing modes Snap to 0

Editing Individual Samples

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

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:
  • Selection: processes the selected portion of the audio file
  • Whole file: processes the whole audio file


• Convert to:
  • Stereo: converts a (mono) audio file to stereo
  • 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.

Snap Selection Edges To 0

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

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.

Looping Audio Files

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.

Save/Recall Selections

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 going to 'view'/'save/recall selections'. Use the timeline selection to select a part.

• 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

Regions

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

Using Beat Doctor

Let's say you just recorded that perfect drum take you want to use in your project, but a couple of drum hits are slightly off-timing. One way to fix this would be to manually edit the recording by slicing the audio region and dragging it to it's desired position, re-aligning it to the project's tempo. Although this is a valid solution, it may get time consuming on very long tracks or when there are many mistakes. n-Track's Beat Doctor provides the tools to quickly and easily fix the timing of your drums, percussion and other rhythmic recordings without having to manually edit the regions.

To use Beat Doctor, right-click on the region you wish to apply it to, and navigate to the Beat doctor menu.

Here you'll find the Detect command, which opens the Beat Doctor settings panel. Beat Doctor works using transient detection. Transients are areas in your audio in which the energy reaches a much higher level over a very short span of time, a behaviour typical of impulsive or percussive sounds. The Beat Doctor panel allows you to specify preferences for detecting transients, as well as choose what actions you would like to perform on them once detected.

On the left of the panel you'll find the detection mode checkboxes. These let you choose the detection algorythm. For most cases dealing with rhythmic recordings, keep the Time detection option checked.

Just below the detection mode you'll find a list of presets that you can choose from to optimize Beat Doctor's performace depending on the type of audio you want to analyze. For example, if your recording has a lot of background noise, you may choose the With background sounds option in the presets menu. The presets will modify Beat Doctor's advanced settings to better analyze your audio. To customize the behaviour yourself, simply open the More options panel.

Once Beat Doctor detects transients on the region, n-Track will add markers at each transient position. On the right of the panel you'll find a list of actions that Beat Doctor can perform for you after the detection.

• Splice to beats
  This option tells n-Track to slice the audio region at every transient position found in the audio.


• Quantize after splice
  Select this option if you wish for n-Track to quantize the regions after having sliced them. The regions will snap to the closest grid position, defined in your grid settings.


• Fill gap after quantize
  If you check this option, n-Track will fill the remaining space left between the quantized regions with what was detected right before or right after the transient. This can be useful to avoid gaps of silence to be left beteen a beat and the next.

These actions are also accesible through the Beat doctor menu, found by right-clicking on an audio region. Another action you'll find at the bottom of the menu is the Apply markers to audio file command. This will write marker position information into the audio's actual .wav file, allowing you to save tempo information along with the audio file.

Types of Effects

Realtime audio effects can be used from within the program in a number of ways:

• Insert Effects


• Aux channel effects


• Group channels


• Master channels


• Live input processing

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.

An Aux channel differs from a Group channel in that the Aux is usually sending a bit of the track's signal 'on the side' to the aux (maintaining unchanged the track current output), while with a Group channel you normally send the whole of the track to the group setting the group as the track's output.

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. See also: Real-time effects VST plug-ins Aux channels routing and settings Signal Path

Live Input Processing

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.

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, Wasapi, 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.

VST Plug-Ins

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.

DirecX Plug-Ins (Windows)

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.”

AU Plug-Ins (Mac)

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.

Effects That Work with n-Track Studio

You can find links to shareware and freeware VST, DirectX and AU effects plug-in on the n-Track Studio website.

Freezing

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/UnFreeze 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-->'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).

Using other programs inside n-Track using ReWire

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

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.
nEfx Compressor, included in n-Track, supports sidechain input.
Third party compressor plug-ins that have support for side-chaining include DensitymkII (free), Voxengo Crunchessor, and FabFilter Pro-C (available as VST3).

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.

Aux Channels and Settings

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:
Iwhen 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.

Group Channels

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.

Automating Effect Parameters

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

- Aux send/return automation

Volumes and Effects Parameters Automation

Click on the Draw Volume Envelopes on the toolbar and choose the desired parameter (track volume, pan, send or return volume etc.) from the track's left panel. On Mac you can also Ctrl+click on any track, select Envelope →, then Select which envelope to draw. 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 on the Envelope Options panel on the track's left panel, and select a parameter form the dropdown list.

By default, n-Track adds a new node each click (i.e. every move of the mouse will create a new node in the piecewise linear waveform).This is useful to manually draw the envelope, while adding nodes one by one is useful when you want more control on the envelope editing. To deactivate this option: right click and activate the Click adds node option. Then, to add a new node, right click and select Add new node from the popup menu, or Shift+Click.

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.

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.

Aux Send/Return 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 Draw Volume Envelopse icon on the toolbar and select "Draw send/return 1 volume" from the track's left panel droprown. 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.

If you are drawing many envelopes, it may be useful to view each automated parameter on a separate track. You can do this by clicking the icon on the track's panel.

Destructive Processing

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:
  • Volume envelope: apply the track’s volume envelope to the selected audio file
  • Pan envelope: apply the track’s pan envelope to the selected audio file
  • Effects: apply the track’s effects to the selected audio file
• Apply to:
  • Selection: process the selected portion of the audio file
  • Whole file: process the whole audio file
• Convert to:
  • Stereo: convert the (mono) audio file to stereo
  • 32-bit: convert the file to 32-bit format
• More Options:
  • Dither: Use dithering
  • Dither depth: Depth in bits of the dithering noise
  • Noise shaping: Use noise shaping

Track EQ Properties

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:

• Track selection: Choose the track you want to analyze.


• Band selection: Choose the band to which you want to change the parameters.


• Band type/eq type selection: Set presetted eq/band type.


• Slope/Band width: Adjust the width of the currently selected band.


• Band frequency: Adjust the frequency of the currently selected band.


• Band Gain: Adjust how much the current band boosts or cuts the signal.

Spectrum Analyzer

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.

Automatic Tuner

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!

n-Track Signal Path

Mastering

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).

Dither & Noise Shaping

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.

Types of Files Used by n-Track Studio

n-Track Studio uses numerous types of files for different purposes:

Track Formats, Mono & Stereo Playback

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.

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.).

You can select the playback mode by clicking on the channel Stereo/Mono state button on the channel mixer stripe:

•   Stereo mode


•   Stereo to Mono mode


•   Left channel to Mono mode


•   Right channel to Mono mode

Packed Song Files (.sgw)

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).

Mixing Down the Final Song

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.

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, M4A or WMA format for distributing the song in a smaller file.

Surround Formats

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.

Creating a Surround DVD

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:

• Audio DVD Creator
• Minnetoka SurCode
• Sonic Foundry Soft Encode
• 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.

Customizing the Program

The colors and fonts used by the program can be configured through the Preferences/Appearance/skins or simply settings/skins

What is Songtree?

Songtree is a free online service that allows to collaborate with other musicians in building songs online. Songtree is developed by the same team behind n-Track.

When someone uploads a song to Songtree, it becomes available for others to contribute to and build upon. Each contribution added by users creates a new version of the song, while the original song remains un-affected. As more people add their contribution, a 'tree' of songs starts to take shape, with each contribution branching out from the original song.

Songtree is available as free web, iOS and Android apps. The apps allow you to record your performances directly from your device, as well as letting you interact with other musicians via comments, likes, invites etc.

Songtree is fully integrated inside n-Track Studio, so that you can overdub other Songtree users songs without leaving n-Track.

Musical Collaboration inside n-Track Studio

If your computer is connected to the internet, you can upload, download, share and interact with musicians from all over the world directly from n-Track Studio.

Say you need an extra musician to play on your song and you just don't know or can't find any, you can upload your song to Songtree directly from your n-Track project, and find one online!

At the same time, there may be many other musicians out there that may need your contribution to their music. Search for songs by genre or instrument, find the ones you like the most and add your part using all the tools available whithin n-Track Studio. If you'll add positive contributions that others will love, you'll quickly build new fans, and your reputation and status whithin the Songtree community will grow rapidly.

For more information about Songtree, visit www.songtr.ee, read the Songtree FAQ, or check out some recent songs created collaboratively by Songtree users.

To start using Songtree whithin n-Track Studio, simply click on the button on the main toolbar.

Exploring the Songtree Community

When you launch Songtree, you'll see a list of songs uploaded by users. You can play and explore the latest or top-rated ones, follow and connect with the artists that created them or take full advantage of Songtree's social features, such as likes, shares and comments.

By clicking on a song, you can view the song's tree, made up of all the contributions linked to the song in question. You can easily see which artists took part in a particular song, see how the collaboration process evolved, and decide at what point of the tree you wish to add your own contribution.

Next to each song you'll find the button, which lets you import the song to n-Track for overdubbing.

Uploading a song to Songtree

To upload a song to Songtree from n-Track Studio, open Songtree by clicking on the button, and then click the Upload button on the Songtree interface. You will be asked to provide a name for the song, as well as other information about it, such as its genre, tempo or key. Once you fill up the upload form and press Upload, your song will be available on Songtree. To see all of your Songtree uploads, simply click on the my Songs button.

Overdubbing a Song on Songtree

Once you found a song you wish to overdub, simply click on the button. n-Track will create a new track for you on which the song will be placed.

You can now start adding your contribution to the song using all of n-Track's features. Once you're done, and you wish to upload the song back to Songtree, simply hit the Upload button. Songtree will automatically detect which track you are recording over, and add your contribution to the song's tree automatically once upload is complete.

Follow Your Songs as they Grow

Songtree will notify you when others overdub, like or comment on one of your songs, as well as when another artists is following you, or when you receive an invitation to overdub a song. You can invite any Songtree user to overdub a song, whether created by you or not. So if you know the perfect musician for contributing to a song, you may invite them even if the song is not yours.

The fun thing about Songtree is to see how a musical idea can take many different directions depending on the artists' choices.

What's Pure Data for n-Track

Pure Data (abbreviated PD) is a visual programming environment for audio, video and graphical processing. This means that you can build audio (or video) processing programs - called patches - connecting basic blocks with virtual cables.

n-Track lets you load effects and synthesizers built as Pure Data patches into a n-Track song, leveraging all the flexibility in sound manipulation of Pure Data, combined with the advanced recording, mixing and MIDI sequencing tools available in n-Track. This is made possible by the LibPd project.

More info on Pure Data can be found here:

• Pure Data official website
• Pure Data Tutorial
• Pure Data on FLOSS Manuals

Why Pure Data for n-Track

Pure Data is very powerful, but integrating sound objects made with it into your creative workflow can be difficult at times. n-Track is here to help with:

• Applying audio effects to complex sequences of recorded clips and tracks
• Allowing synthesizers to play musical scores composed with either the piano roll or the step sequencer views
• Saving the parameters of an effect or a synthesizer
• Automating the parameters evolution in time
• Visualizing the patch’s controls only, to focus on the performance

Creating a Pure Data effect patch for use in n-Track

Open Pure Data, create a new empty patch, and add both an [adc~] and a [dac~] object. They are needed to let n-Track know the audio entry and exit points.

Build your patch, adding and connecting the blocks you need to modify the audio signal as you desire.

Please note that you need to keep the left and right channel path separated if you want to keep your audio in stereo. A good practice is duplicating the audio processing blocks and controlling “twin” blocks with the same gui control:

That’s all! You can load this patch in n-Track, and apply this effect to any track. [see also: using a n-track pd patch]

Creating a Pure Data instrument patch for use in n-Track

Open Pure Data, create a new empty patch and add both a [midiin] and a [dac~] object.

The [notein] object will let the patch receive MIDI notes from n-Track’s sequencer tracks. You can also add a [ctlin] or [midiin] object to receive any kind of MIDI event from n-Track. The [dac~] object allows n-Track to catch the synthesizer’s audio output.

Then add any object you need to generate sound and control it

A good practice to ease the reuse of the sound generator core is to put it into a separate patch file (called “abstraction”).

To build a polyphonic synth, you can just load the sound generation abstraction multiple times. You can distribute midi notes to the various voices using [poly].

Pure Data objects supported by n-Track

Pure Data for n-Track currently supports all Pure Data (vanilla) objects, plus [fiddle~] and [expr~].

The GUI controls window is currently able to display:

• Horizontal and vertical sliders
• Horizontal and vertical radio buttons
• Toggles
• Bang Buttons
• Canvases
• Comments

Guidelines and known limitations

There are a few things to keep in mind when building Pure Data patches to be used in n-Track:

• Currently, only two inputs and two outputs are supported, any additional channel will be ignored by n-Track.
• Only GUI objects placed in the main patch will be shown by n-Track. All GUI objects in abstractions or sub-patches will be ignored, even if they are visible in the main patch through the “Graph on parent” mechanism.
• When loading the patch, n-Track will parse it and generate a temporary patch with some additional informations, needed to bind to the supported GUI objects. This means that if you modify the patch using Pure Data, you will have to reload it in n-Track.
• n-Track should be able to load any external built for your platform and placed in the same directory as the main patch, but this feature is still experimental.
• Patches to be loaded in n-Track must avoid using the [dsp~] switch, since this is currently causing issues within the engine.

Load an effect patch

Right-click on an audio track and select “Effect → Add Effect → VST3 → Pure Data → n-Track Pd [Fx]”. Click on the button, and select Load patch.

Navigate to the location of the patch you want to load, and select it.

Load an instrument patch

Create a new instrument track: from the “Add Channel” menu, select “New instrument channel → VSTi → n-Track → n-Track Pd [Instr]”. Click on the button, and select Load patch.

Navigate to the location on disk of the patch you want to load, and select it.

To get MIDI messages, select the instrument as the destination in a MIDI or step sequencer track.

Load a mixed effect/synth patch

Pure Data for n-Track is able to receive MIDI events when added as an effect to an audio track, but it’s not able to receive audio when loaded as an Instrument. For this reason, if your patch is both processing incoming audio and listening for midi events you should load it as an effect on an audio track.

Please note that loading a synthesizer patch with no audio input on an effect track will break the audio flow from n-Track’s audio track to the mixer.

GUI modes

Pure data for n-Track can show your patch’s control using Pure Data's graphical style, or a custom style designed for n-Track. You can switch between the two skins from the menu, toggling the "User Pure Data Skin" option.

Load and save the patch's state in n-Track

You can save the current state of the patch as a preset, clicking on the button and typing a name for the new preset.

You can then load the saved patch at any moment, clicking on the drop-down menu and selecting one of the available patches. Your custom saved patched will appear below the default ones offered by n-Track.

Automation of GUI-controlled parameters

Pure Data for n-Track allows the automated control of the patch’s parameters exposed as GUI objects. To automate a parameter, enable automations clicking on the button, then click on the automation selector in the audio track’s panel.

Click on “Effect parameters → Add parameter…”, then select the “n-Track Pd” plugin and the desired parameter from the “Effects parameters envelopers” window.

Pure Data console in n-Track

You can show n-Track’s pure data console for debugging purposes, clicking on the “Show console” option in the menu.

Introduction

The step sequencer lets you compose rhythms and songs using smaller pieces - called patterns - as the building blocks for more complex scores.

To open the step sequencer window click on the button in n-Track’s toolbar, or double-click on a step-sequencer track in the timeline.

To create a new step sequencer track click on “Add new empty track → Step-Sequencer” from the “Channel” menu.

Freerun and Playlist modes

Each step sequencer track can operate in one of two modes:

• In freerun mode the currently displayed pattern will be played indefinitely in loop. This mode is useful for testing ideas, practicing, jamming and live performing.
• In playlist mode you can assemble a customized sequence of patterns. This is especially useful for composing songs with different sections (eg: verses, choruses, fills).

You can switch between the two modes at any moment, using the mode selector.

Select the destination instrument

The step sequencer track outputs midi notes to any MIDI-capable instrument and plugin. You can select the destination from the step sequencer track’s mixer stripe, or from the track’s side panel.

Creating a free-running pattern

While in freerun mode, create a new pattern in the pattern roster, clicking on the button. The pattern roster contains all the available patterns for the track, you can think of it as a toolbox filled with rhythms ready to be used.

After creating several patterns in the roster, you can switch among them clicking on the desired one. If the song is playing, you will immediately hear the new pattern right after selecting it.

Adding notes to a pattern

To add notes to the currently selected pattern, click on a cell in the pattern’s note grid. The cell will light up and the corresponding note will play when the progress cursor crosses it.

The note for the cell’s row is indicated in the note list, on the left side of the grid.

Click on the note’s button and drag the mouse vertically to modify its velocity. The button’s light will dim reflecting the velocity level.

Click on an active cell to deactivate it.

Creating a playlist

Switch to playlist mode: from now on, patterns in the roster will not be played when you create or select one of them. Only patterns added to the playlist will play in sequence.

To add a pattern to the playlist, click on the button while the desired pattern is selected, or drag the desired pattern from the roster to the playlist.

You can arrange the patterns’ ordering in the playlist at any moment, dragging them where desired.

When playback is activated, the playlist will be reproduced, following the main song time position.

To remove a pattern from the playlist, select it and click the button. The removed pattern will still be available in the roster for reuse.

Setup pattern properties

The main settings display shows the most important properties of the selected pattern

• Pattern: shows the pattern’s name. Click on the name to modify it
• Steps: shows the number of time steps for the selected pattern. Click on the arrows to change it
• Step Length: indicates the time distance between each step, and the base duration of each step (the duration can be modified for each step independently, see [Notes duration]. The length is expressed in musical notation
• Track: indicates the track you are working on. Click to select a different track.
• Bpm: indicates the play speed in BPM. This is a global setting (not specific to the pattern), mirroring the one in the main toolbar.

Permanently delete a pattern

To permanently delete a pattern, click on the button while the pattern is selected. When a pattern is removed from the roster, it will be permanently removed from the playlist as well, if present there.

Clone a pattern

Clicking on the button, the currently selected pattern is copied and a new identical pattern is added to the roster. This is useful to start working on a new pattern that is a variation of a previous one. The modifications performed on the new pattern will not affect the original one.

Notes duration

By default, each note has the same length as the time separation between note-on events. (see: Step Length).

If you need a note to be played for a longer or shorter time, adjust step length by dragging the right hand side of a step. Dragging right will increase step duration, dragging left will shorten step duration.

A few things to keep in mind:

• The duration of the note is proportional to the step length
• The selected duration will affect all the cells belonging to the same column
• lengthening or shortening a step will not affect the position in time of the following step

Customize notes list

Each pattern’s note grid can have a customized number of rows, each one representing a pitch value. To add or remove rows from the pitch list, click on the buttons.

You can transpose the note list - and thus the pattern - clicking on the buttons, and you can even set a custom note for each row, clicking on the button and selecting the note from the resulting menu. From the same menu you can select one of the alternate namings available for the note list

You can also load specific scales in the midi list, to ease the composition process. Just click on the scale button, and select a major, minor or other type of scale in the desired key from the menu. Click on Advanced to fully control all aspects of the scale generation.

Pattern color

To change the color of a pattern, right-click on it in the roster and select the desired color from the Color sub-menu.

Export/Import playlists and patterns

To export all the step-sequencer playlist to a file, right-click on any pattern in the playlist and select “Export Step-Sequencer playlist”, then select a name and a destination for the saved file.

This can be useful if you want to reuse the same patterns in another channel of the same song or in another song. To do this, right-click on the destination playlist, select “Import Step-Sequencer playlist” and select a previously exported “.nspl” playlist file.

Please note that all patterns already present in the track’s playlist and roster will be replaced by the imported ones!

You can also export a single pattern or an entire playlist to a MIDI file for use in a MIDI track or in an external MIDI sequencer. To do that, right-click on a pattern in the playlist and select:

• "Export pattern to MIDI file", to save a single pattern (also available when right-clicking on a pattern in the roster)
• "Export playlist to MIDI file", to save the entire playlist.

Then select a name and a location for the destination MIDI file.

Select a pattern using a MIDI device

You can associate each pattern in the roster with a MIDI message, to quickly select it in realtime using a MIDI controller. This is especially useful in freerun mode, to switch the playing pattern while jamming or performing.

To assign a MIDI event to a patter, right-click on it in the roster, and select an option from the "Associate to MIDI control submenu. You can even use n-Track's MIDI learn ability to perform the association."

Step sequencer track properties

The step-sequencer track can be considered a special type MIDI track. For this reason, a step-sequencer track’s properties are the same as a MIDI track's properties.

Editing Multiple Tracks on a Single Piano-roll

The Piano-roll window optionally allows to view and edit multiple MIDI tracks. By default the Piano-roll opens in Track mode. A column on the right shows a button for each track number. Click a track’s button to switch the Piano-roll to that track.

Click on the Song mode button to show events from all MIDI tracks. When you are in Song mode you will still edit (i.e. add, move and delete events) on a single track at a time, but you will see MIDI events from all the tracks.

To switch which track you want to edit click on the track number button on the left side of the right track browser pane. Click on the button to the right of the track number to hide or show a MIDI track.

Locating the Samplers

The sampler types can be selected in the instrument browser under ‘My instruments’ -> ‘Create new instruments’.

Melodic Sampler

Getting Started with the Melodic Sampler

Drag samples or one-shots from the loop browser into the sampler and start playing.

Drag another sample into the melodic sampler to update the sound.

The sampler will automatically recognize if a loop has root note information and will use it to pitch the sample correctly. If the sample has no root note information it can be set with this control. Changing the root note can also transpose the sample.

Change the range of notes that the sample will cover by dragging the yellow range control on top of the keyboard. When a sound is dragged the default range is 2 octaves but can easily be extended using the range control.

Recording in the sampler

The rec button lets us quickly record our own samples. It’s surprising how easy it is to record something that can work as a playable instrument.

Use the fade controls at the beginning and end of the sample to soften the attack and release of the sample, while the proper ADSR envelope that controls how the sample volume is modulated as a note is pressed, held and then released can also be edited.

Additional controls

The pitch knob can be used to alter the pitch.

Use the reverse command to invert the playback direction of the sample to create interesting effects.

Play, Edit, Parameters views

The sampler view has three pages. Enter the edit page with the pencil icon; the keyboard icon goes to Play mode, where more n-Track or 3rd party effects can be added, as well as the arpeggiator. The ‘Sliders’ icon shows the parameters page to get into the nitty gritty details of the sampler. Press the XY button to view all the available parameter pages.

Drum Sampler

Getting Started with the Drum Sampler

Selecting ‘Drum Sampler’ from My Instruments, sounds can be dropped onto the pads to build custom drum kits.

Toggle between the Pad and Keyboard views using the Pad icon, located above the Sampler keyboard.

Mute groups

One interesting feature of the drum sampler is the Mute group. Sometimes it's useful to have a sample stop another sample from playing. A classic example is the closed and open hi-hat. The open hi-hat usually has a long ringing sound, and when the drummer releases the pedal the hi-hat stops ringing. This can be accomplished by assigning the closed and open hi-hat to the same mute group. When a sample in say mute group 1 starts playing, it will stop any other sample belonging to the same mute group.

Playing Screen Drums with Sampler

To play drums on the screen, switch to the full screen-drums view.

Slicer

Import a sample to get an automatically sliced version of the sample mapped to the sampler keyboard.

Slice by Grid

Choose how to slice the sample to the keys by selecting the Slicer dropdown and, using the ‘Slice by’ Grid setting, choose a subdivision and tempo.

Slice by Transient

‘Slice by’ transient can be selected, as well as control detection sensitivity and the style – from percussive to melodic.

Add and remove sliced regions

After the slicer has performed the automatic slicing, the individual sliced regions can always be adjusted manually. It's also possble to add and remove sliced regions.

Multilayer Sampler

With the Multilayer Sampler you can add multiple samples (‘layers’) on a single note. These layers can be individually edited to create more dynamic and complex instruments. The multilayer sampler is included in the Extended and Suite editions of n-Track Studio.

'Upgrade a Sampler type to Mulilayer'

You can turn or ‘upgrade’ a track with any of the other sampler types into a multilayer sampler track, by selecting the Multilayer sampler in the instrument browser. Note that this doesn’t work the other way around: if you have a track that already has the Multilayer sampler and select for example the Melodic sampler, you’ll get an empty Melodic sampler. So you can always start building your instrument from one of the more intuitive and easy samplers, and then upgrade to the Multilayer sampler when you want to get more options and freedom to build your instrument.

Working with layers

Import a sample, and enable the “new layer” icon, so that when a second sample is dropped onto the selected note it will be added to the note, without replacing the existing sample.

Scroll between the layers using the dropdown menu and listen to a layer individually by activating the ‘S’ solo button then playing the associated note.

Each layer can be tuned by assigning the root key or, for quick adjustments, using the pitch knob.

Importing a sampler with the new layer icon deselected will replace all of the layers in the multilayer sampler. If we happen to do this by mistake we can always undo and return to the previous state.

Trigger samples by velocity

Sometimes you may want to have a note always play multiple samples, while sometimes you may want to have a sample play only when you press the note in a “soft” way (i.e. with low velocity) and another sample play only when the note is pressed hard. This can be accomplished by having two (or more) layers assigned to the same note with velocity ranges that don’t overlap. One layer can, for example, have the range 0 to 50% and another 51% to 100%. So when the key is pressed with a lower velocity it triggers one samples, and when the key is pressed with a higher velocity it triggers another sample. Note that on the screen keyboard the strength/velocity of hitting a key can be simulated by tapping on the upper (soft) or lower (hard) part of the key.

Duplicate layers

A layer can be duplicated with the ‘Clone sample’ icon. The duplicate will be added to the multilayer dropdown. Once a sample is duplicated its settings can be changed independently from the original layer.

Delete layers

A sampler layer can be removed by clicking the trash icon.

Loop Modes

Select the Loop mode icon to adjust the current sample loop mode.

"One Shot" mode

“One Shot” is the default setting and plays the full waveform to completion.

"Loop" mode

With the ‘Loop’ mode selected we can select loop points that we want to assign to the sample. The sample cycles between the loop points until the key is released. Note that is the sample clicks when cycling between the set loop points, there is a fade function that can be utilized to prevent this, or to create creative crossfades between the start and end of a sample.

"Loop until release" mode

“Loop until release” works similarly to the "Loop" mode, with the difference that when the key is released, the sample stops looping and continues to play beyond the loop end point to the end of the imported sample.

Resampling Mode

This menu sets the resampling algorithm used to change the pitch of a sample to match the key that is pressed on the keyboard. Let’s review the options:

Speed follows pitch

When ‘Speed follows pitch’ is selected the speed of the sample increases or decreases together with the pitch. This makes higher pitched sounds play shorter than lower pitched ones. It works very well for solo instruments but may not be appropriate for chords because the various keys of the low and high pitches will have different speeds and may sound out of phase.

Maintain original speed

‘Maintain original speed’ is the default setting and keeps the sample length consistent across the assigned key range.

Preserve transients

‘Preserve transients’ maintains the sample speed regardless of the pitch. It may often sound better than ‘maintain original speed’, although setting up the sample after dragging may take some more time as the sampler processes the various pitches in the range.

Preserve formants

‘Preserve formants’ is similar to ‘preserve transients’ but also keeps the characteristic frequency components of the sample, keeping an instrument or vocal sample sounding as it was produced by the same instrument but at a different pitch, as opposed to sounding like it was played by a different instrument or singer. For example this mode avoids the classic ‘chipmunk’ effect when pitching vocal samples higher.

Saving custom instruments

When we’re happy with a custom instrument, we can save it as a preset using the save icon in the upper toolbar.

The instrument can be easily recalled from the Instrument Browser, under ‘My Instruments’.

Custom Sound Import

We can edit the existing sound, record a new sound or import our own custom sound in different views.

The edit menu let’s us Rec, Import or Edit sounds in Pianoroll, Step Sequencer, Screen Drums and Pad view. Super useful for catching inspiration on the go, or refining sounds back in the studio.

About instrument Plug-Ins

Both the instrument plug-in’s audio channel track and the MIDI track’s volumes and pan settings can influence the signal.

Loading instrument plug-ins

The instrument browser lets you choose the instrument you want to play via MIDI. It will popup when creating instrument tracks - or when you click on a MIDI track’s output slot - so you can easily choose a new instrument or switch the sound on an already created instrument track. The browser is organized in three sections:

• External instruments: Here you’ll find all supported third-party virtual instrument plugins available on your system.
• n-Track instruments: In this section, you’ll find a list of n-Track’s built-in instruments. Instruments and patches which show the icon require an additional download, which you can launch via the Add-on manager.
• MIDI outputs: shows the available hardware MIDI outputs, as well as already created instrument channels which you can send MIDI to, allowing you to route a new MIDI track to an instrument you already created.

When you select an instrument from the instrument browser, n-Track will create a new instrument channel with the instrument loaded up, and assign it to a MIDI track. To start writing MIDI, just click on the icon to open the piano roll window.

In this example, ‘Piano’ was chosen as a virtual instrument, as you can see from the MIDI output slot marked with O. If you have MIDI controller plugged in, make sure this is selected as MIDI input in the I slot, or that the default All channels, all devices is selected.

To change instrument, simply click on the output slot, which will re-open the instrument browser.

The Add-on Manager

The Add-on manager lets you download free and premium instruments, sounds and drumkits available for use in n-Track.

In the Add-on Manager you'll find:

• Free instruments: You’ll find a collection of virtual instruments which are free to use for n-Track users, and that only require to be downloaded in order to be used. Clicking on the ‘Get’ button will automatically download and install the extra content, and the instruments will be available in the instrument browser next time you open it.
• Premium instruments: Some instruments require to be purchased. To buy an instrument, simply click on the ‘Buy’ button on the instrument you desire. You’ll be directed to n-Track’s Add-on Store, where you can complete your payment. Once you buy the product, your sounds will be automatically downloaded and installed, and will be available for use next time you open the instrument browser.

Playing Live Through an Instrument Plug-In

Instrument plug-ins can be used to play a MIDI instrument live (i.e. let the instrument plugin 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 Settings/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 Wasapi, 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.

About SMPTE/MIDI Time Code

The MIDI Time Code (MTC) is a protocol used to exchange SMPTE (Society of Motion Picture and Television Engineers) time signals over a MIDI connection. The SMPTE timing is the standard used to sync different machines such as video tapes decks, multitrack recorders, etc. in both the professional audio and video world. The standard SMPTE signal coming from hardware devices can be converted to MTC using dedicated interfaces.

Many software packages support MTC sync, and the syncing of different programs in the same computer can be accomplished without having to resort to hardware cables. It's possible to use a loopback MIDI driver that simply routes the MIDI signals sent to its inputs back to its virtual MIDI output ports.

You can download freeware utilities that accomplish this task.

In this way, it's even possible to sync multiple instances of n-Track on the same computer or even in different computers connected by a MIDI cable!

About MIDI Clock

MIDI Clock is a protocol used to sync different MIDI devices to the same clock reference. A program or device that generates a MIDI Clock signal sends a “MIDI clock” impulse every 1/24 of a quarter note. In this way, unlike MTC sync, in which absolute time is the only information sent, syncing to MIDI Clock automatically allows the slave to follow the master tempo variations.

Apart from the clock signal itself, three other signals are exchanged:

• Song Position Pointer: sent to signal that the playback will be restarted from a new location


• MIDI Continue: sent when the playback is being restarted


• MIDI Start: sent when the playback is being started from the beginning of the song

Many software packages support MIDI Clock sync, and the syncing of different programs in the same computer can be accomplished without having to resort to hardware cables. It's possible to use a loopback MIDI driver that simply routes the MIDI signals sent to its inputs back to its virtual MIDI output ports.

You can download freeware utilities that accomplish this task. In this way, it's even possible to sync multiple instances of n-Tracks in the same computer or even in different computers connected by a MIDI cable!

SMPTE Time Formats

The time format most commonly used between audio programs is 30 fps non-drop. To make n-Track generate this timing, select "30 fps non-drop" from the MTC/MIDI Clock dialog box (accessed by selecting Settings/Synch with other devices), enable the "use real 30 fps for 30 fps non-drop" checkbox and click on the “Enable” button.

This is all you really need to know. However, if you are willing to dive into the subtleties of time formats, here’s some more info:

There are 5 SMPTE time formats that can be used with MTC:

• 24 frames per second (used mainly for film sync)


• 25 frames per second (as used by the European television standard)


• 29.97 frames per second with drop frame (also referred to as 30 fps drop)


• 29.97 frames per second (also referred to as 30 fps non-drop)


• 30 frames per second (used only in the audio world)

The last two formats are used to sync to NTSC video, which has a frame rate of exactly 29.97 fps.

The seconds counter is always incremented when the frame count reaches 30, so if the frame rate is 29.97, the SMPTE time soon departs from the real-world time. For example, when the SMPTE 29.97 fps non-drop time is exactly 1:00:00.00, the real world time would be 1:00:03.18, 108 frames higher than the non-drop time.

The 29.97 fps "drop" time format is used to compensate for this problem. In this format, two frames are dropped every minute, i.e. the time advances from 0:59:29 directly to 1:00.2 for each minute in the hour except at minutes 0, 10, 20, 30, 40 and 50. In this way, a total of 108 frames are dropped every hour, thus perfectly compensating for the "slowness" of the 29.97 frame rate. Note that this leads to some times that are never used, such as 1:00.00, 1:00.01, 2:00.00, and so on.

Configuring the Program to Act as Master

Now, every time the playback is started from inside n-Track, the device the program is synced to will (after a short delay) start the playback.

Configuring the Slave to an Incoming Time Code Signal

Open the Settings/Preferences/MIDI/MIDI devices dialog box, then set the MIDI input port (for example, LB1 if you're using a HUBI Loopback device) from which you want to receive the time code signal.

Open the Synchronization dialog box by selecting the View/Synch with other devices menu. Click on the “Enable” button corresponding to MIDI Clock or MTC. On the program or device you want to sync, enable master MTC or MIDI clock syncing and select the output port corresponding to the output port previously chosen.

Now when the playback is started from inside the master device (the device that is generating the sync signal), n-Track will also start playback. When the record button is pressed and slave syncing is enabled, n-Track will start the recording as soon as it receives the incoming sync signal.

n-Track needs a few instants to prepare itself for playback: after detecting an incoming signal, the program will actually start the playback after the time specified in the "pre-roll" field (the playback will start with this delay, but it will be nonetheless in sync with the other device). The program will stop the playback after the time specified in the free-wheel field has elapsed without having received an MTC signal.

You can compensate for possible sync problems by changing the "offset" and "set 0 at" values in the Synchronization dialog box.

MTC/MIDI Clock Dialog Box

You can configure additional synchronization options in the MTC/MIDI Clock dialog box:

MTC/SMPTE output:


• Output port: port to which the MTC signal is sent to


• Enable: enable MTC generation


• Format: SMPTE time format used for the MTC generation


• Offset: offset that will be added to the generated MTC (this value will be summed to the time you see on n-Track's time window. If, for example, the offset is 5 seconds, when you start the playback within n-Track the first timecode generated will be 00:00:05.00)

MIDI Clock output:


• Output port: port to which the MIDI Clock signal is sent to


• Enable: enable MIDI Clock generation


• Offset: offset that will be added to the generated MIDI Clock


• Use Start, never Continue: always send the MIDI Start signal instead of MIDI Continue


• Send Song Position Pointer: send the SPP signal to notify the slave of the new position within the song


• Wait after sending SPP: waiting to allow the slave to cue to the requested position


• Use internal timer: use the internal timer instead of the soundcard's clock reference (recommended). The internal timer resolution is set through the Settings/Preferences/MIDI dialog box


• Use real 30 fps for 30 fps non-drop: use the true 30 fps frame rate instead of the 29.97 NTSC frame rate (see SMPTE Time Formats)

Input:


• Input Port: port from which the program received MTC or MIDI Clock signal


• MTC/SMPTE - MIDI Clock: select the type of the incoming sync signal


• Enable: enable syncing to the incoming signal


• Set 0 at: offset in the song at which the playback is started


• Start at offset: offset in the incoming SMPTE time at which the program will start the playback. For example, if the "start at offset" value is 1 hour and "set 0 at" value is 5 minutes, the program will start the playback when the SMPTE it receives reaches 01:00:00.00 and the playback will start at the offset 00:05:00.00 in the song (MTC only).


• Preroll: time that the program takes to actually start the playback after it has detected from the MTC input that the playback should be started


• Free Wheel: after not receiving a valid MTC signal for this amount of time, the playback will be stopped (MTC only).

Virtual MIDI Patch Cables (Loopback Devices)

A Virtual MIDI Patch Cable or Loopback Device is a driver that generates virtual MIDI input and output ports. Every MIDI event sent to a loopback output port instantaneously appears at the corresponding input port. The driver also allows routing the same output to more than one input. This can be used to sync n-Track Studio to one or more audio programs using MIDI time code, without having to resort to external MIDI connections.

You can download freeware virtual MIDI patch cables utilities from the n-Track studio website’s download section.

Syncing a Video Clip to Playback

n-Track Studio can also synchronize a video file to a song’s playback.

From now on, every time the playback is started, the video file will be played back in sync with the program's audio output. When not in playback, a single click on a given point on the timeline time slider will cause the corresponding video frame to be displayed in the Video Sync widow.

The program will be able to play any file type for which a proper codec is installed on the system. You can check which codecs you have installed in the Control Panel/Multimedia/Devices/Video codecs dialog box. Windows usually has codecs for playing .AVI, .MPG and .MOV files installed by default.

Video playback can use a great quantity of processing power, depending on the size of the video window: to reduce the load on the computer, reduce the size of the Video Sync window.

If you are experiencing playback problems you can download free filters, codecs and tools online that let you play popular and not-so-popular audio and video formats. K-Lite Codec Pack is one example.

File Menu

The File menu lets you work with song and audio files in n-Track Studio.

Edit Menu

The Edit menu contains commands for altering waveforms in n-Track.

Track Menu

The Track menu allows you to work with entire audio tracks.

Transport Menu

The Transport menu controls audio playback and recording.

Add Channel Menu

The Add Channel menu allows you to insert new channels into the multitrack environment.

View Menu

The View menu contains options for customizing the program’s interface.

Settings Menu

The Settings menu allows you to configure the program.

Help Menu

n-Track’s Help menu offers assistance with the program’s operation.

Recording Settings

Options Settings

Appearance Settings

MIDI Preferences

Paths Settings

Audio Devices (Advanced)

n-Track supports different types of soundcards drivers. To open the audio Devices dialog box, select Settings/Audio devices and click “Advanced.”

Asio Settings

To open the Asio dialog box, click the “Settings” button on the VU meter, then click “Asio Settings.”

Buffering Settings

To open the Buffering Settings dialog box, select Settings/Buffering settings. Click “More Options” to open the buffering options.

Click the “Apply” button to use the new values. If playback, recording or live input processing are active, they will be stopped and restarted with the new buffering settings. The default values appear when you rotate the knobs, but any other value for both the buffer size and number can be manually entered typing the number in the text boxes.

Always start from the default values, then try to increment the buffers if you need more reliable playback and decrement them if you want lower latency.

When using Asio drivers, the buffer size and numbers are dictated by the Asio driver, so the program will complain if the settings are different from those requested by the driver.

MIDI Devices Settings

MIDI Track Properties

Right-click a MIDI track, then select “Properties” to open this dialog box.

MIDI Event Properties

Right-click a MIDI event in the Piano Roll, then select “Properties”:

MIDI Events Editing Window

See also:
Editing MIDI tracks

The MIDI events editing window allows you to sequentially view and edit all the events in a MIDI track in a table that shows all of the event’s parameters. For example, for a note event, the note name, channel, start time, velocity and duration parameters are listed in the event’s row in the table. Some events (for example, program changes) may be easier to add or edit using this table view instead of the piano roll view. As with the Piano Roll, you can customize which events are displayed and which can be inserted in the dialog box that appears when you select Settings/Preferences/ MIDI/MIDI events shown in the timeline.

Piano Roll Settings

The piano roll window can show a number of different MIDI events. In this dialog box, you can select which events to show and which kind of event to insert when the button is activated. Click the button to open the Piano roll settings:

Place	select which event to place when the button is activated.
View	select which event should be drawn on the timeline window.

Mixdown Dialog Box

Select File/Mixdown song, then click the “More Options” button to see the mixdown options.

The two typical applications of the mixdown feature are:

• Mixing down the whole song to a single WAV file
• Mixing down some tracks to apply effects permanently or to reduce the number of tracks that the program mixes “on-the-fly”

For the first application, typically all the tracks will be selected (as the program does by default) and the “Process Master channel” option will be checked For the second application, the “Process Master channel” option will be typically unchecked, and “Substitute tracks with the mixed down file” will be checked. In this way, the program will create a new audio file with the selected tracks and will substitute the tracks with this single audio file, in such a way that the song will sound exactly the same as before but the actual number of tracks will be decreased, thus making the program occupy fewer system resources. The program will also ask for a .sng filename to which to export the mixed down tracks, so that, at a later time, it will be possible to reload the single tracks as they were before mixing down (audio files associated with the tracks should obviously not be deleted) using the “Track/Import tracks from .sng file” menu option.

Metronome

Open the metronome settings window by clicking on the metronome icon on the toolbar. Set the tempo, choose MIDI or PC speaker output and choose whether the metronome should be hearable during recording and/or during playback. Click on the test button to hear if the tempo is right.

Time Stretching

This option allows you to change the speed and size of an audio file. To stretch an audio file, click on its waveform in the timeline, then use the Track/Special/Time Stretch menu command.

Fade Properties

To open the Fade Properties dialog box, select Edit/Volume-pan drawing/Fade settings.

WAV to MP3 Conversion

MP3 (which stands for MPEG Layer 3) is a standard compressed audio format that guarantees high sound quality while obtaining very high compression ratios. Depending on the chosen bit rate, compression ratios up to 1:10 can be achieved with an almost inaudible (at least to a non-audiophile’s ears) loss of quality. This makes this format ideal for distributing audio files over the Internet or in any situation when the size of regular WAV files will make their transmission impractical.

On the other hand, MP3 is always a lossy compression, so it isn’t suited for storing audio data when the sound quality must be kept as high as possible. For this reason (and also because on-the-fly decompression of MP3 files will be way too heavy on the computer’s CPU) n-Track Studio doesn’t support MP3 file format for recording or importing tracks. It only allows conversion from finished WAV files (i.e. created after mixing down) to MP3. To listen to the MP3 file you’ll need an MP3 player, such as Winamp.

For the actual MP3 encoding, n-Track uses the free, open-source LAME encoder. You can download the LAME encoder from the n-Track website; the first time you run File/Convert WAV file/To MP3, the program will ask you to specify the location of the file “Lameenc.dll.”

Determine the sound quality of the converted MP3 by changing the LAME “Mode” options using the drop-down menu:

WAV to WMA Conversion

Converting WAV files to WMA format is useful for distributing audio files over the Internet or in any situation when the size of regular WAV files will make their transmission impractical. WMA files contain audio data that is compressed using a high-quality lossy audio compression scheme (codec) called the Windows Media™ Audio codec. WMA files can be significantly smaller than MP3 files of comparable audio quality.

To convert to WMA, select File/Convert WAV file/To WMA (Windows Media Audio), then set the encoding options:

Multiple conversions can be executed simultaneously by the program, and normal program operation can continue during the conversion(s).>

Playing WMA Files
A number of players support the WMA standard, including Winamp v2.5 and later and the newest Windows Media Player. If you attempt to play a WMA file in an older version of Windows Media Player, the player will recognize the new file type and will automatically download the appropriate codec from the Internet.

Ogg Vorbis Conversion

Ogg Vorbis is a non-proprietary standard for audio file compression. Although not as popular as MP3, it is rapidly gaining acceptance and support from audio software, as it’s currently considered the best audio format in terms of achieving the best sound quality for a given file size. Most current audio players, such as Winamp, can play .ogg files. More information on Ogg Vorbis can be found at Xiph.org. n-Track Studio can convert WAV files to the Ogg Vorbis format with the File/Convert WAV file to/Ogg Vorbis menu command. It also uses Ogg Vorbis for compressing audio data saved in packed song files (.sgw), and can import .ogg files into the current project using the Import WAV file command in the Track menu.

Sampling Frequency Conversion

This option allows changing the sampling rate of audio files; you can run this conversion by selecting File/Convert WAV file/Change sampling frequency. The program will also automatically ask to convert the sampling frequency if you try to import a WAV file with a sampling frequency different from the working frequency (as set in the Settings/Preferences/Recording settings dialog box), or if, when doing a mixdown, you select a sampling frequency different from the working frequency.

Recovering a WAV File from Raw Audio Data

Sometimes WAV files get corrupted following hardware (hard disk) or software (file system, Windows crash, program crash etc.) failure. The File/Convert WAV file/Recover .wav file from raw audio data command allows you to try recovering the audio data from a WAV file whose headers (i.e. the beginning of the file which contains information about the file format) have been somehow corrupted.

In order to restore the WAV file headers, n-Track will ask you to manually specify the format that the wav file uses, including the file’s sampling frequency, sample format (16 bit, 24-bit, etc.) and number of channels. If you don’t know or remember which format the WAV file is supposed to have, you can try going by trial and error. Start trying each of the bit formats. If you select an incorrect bit format, the resulting WAV file will play as pure digital noise. If the recovered WAV file resulting is too slow or too fast, an incorrect sampling frequency or number of channels might have been specified.

The Audio data starts N bytes from the beginning of file parameter should typically be left at the default setting, particularly for files originally created by n-Track. If you change the setting, you should change it by a multiple of the file sample size. For example, a 16-bit mono WAV file has a sample size of 2 bytes. Increasing the Audio data starts… parameter by 1 will result in all the file samples being shifted by 1 byte, resulting in pure noise. Likewise, if the format is 16-bit stereo (2 channels), shifting by 2 bytes will result in the file’s left and right channels being exchanged.