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 - 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 - 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.

WaveRT drivers are currently available 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.

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.

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.