The new Bass Amp plugin features bass amp head simulation, and you can choose between various cabinet models, for sounds ranging from deep, driven basses to more present and high-end rich ones.
Check out the demos for some examples of what the plugin can do to a clean, unprocessed bass sound. The new VocalTune plugin lets you easily fix your vocals pitch. You can also snap your vocals to common scales, as well as create your own. Using VocalTune you can do anything from simple pitch correction to completely transform your vocal. Check out the demos for hearing some of the results you can get using this plugin.
Write and edit MIDI data for controlling virtual instruments using the intuitive built-in piano roll editor. Create beats, sequences or arpeggios easily with the new Step Sequencer.
Start a song in one click using the factory patterns, or have fun creating your own. The new screen Drumkit controller lets you play drum sounds with the look and feel of a real kit. The new integrated Songtree app lets you make music with others online. Get another musician into your project, or contribute to a song started by others. You can now load your Pure Data patches inside n-Track and use your favorite sounds and effects in your song. The included n-Vocals plugin offers useful and creative ways of tweaking and manipulating your vocal recordings.
Supports surround mixing for creating DVD audio projects using 5. 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.
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.
Suppose you want the echo to be applied only to the final part of the second solo. We can obtain this using aux send or return automation:. 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. 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. CPU power than putting a separate reverb on each track, and will also make the two vocal tracks blend together better. The next step in mixing is to refine the volume of the tracks during the evolution of the song.
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. 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.
Select "Master volume" on the popup menu that appears when you click on the volume icon on the toolbar. Now that everything sounds good, the last step is to mix down all the tracks to a single audio file.
Select the "Mixdown Song" command from the File menu, then type in the destination filename and click "Save. 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 "" 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. Listen to the track to hear how your mixdown sounds. You may now need to cut unnecessary silent lead-in and lead-out parts in the mixdown audio file. The last optional step is to burn the song to a CD-R. Import the mixdown audio file you just created, insert a blank CD and press the "Burn" button.
You can also tell the program to mixdown and automatically burn the track to an audio CD by selecting the "Burn audio CD track" option in the mixdown dialog box. Good luck with your next song! This chapter contains information regarding the most common tasks that need to be performed when recording or mixing songs using n-Track Studio.
To make your work with n-Track Studio as effective as possible, please take the time to read this chapter. When you launch n-Track, you have various options to quickly start working on your song:. 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. 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. Levels are measured from the maximum level downward. For example, dB is almost perfect silence, dB is barely audible and -6 dB is a very strong signal.
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. 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.
The signals inside n-Track's virtual mixer are floating point audio signals, which support levels far above 0 dB. 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. Learn more about selecting audio devices in the Audio devices tutorial video Windows. 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. Exclusive mode is however recommended for pro-audio. 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 bit version on the x64 version of Vista, low buffering settings result in an error when opening the soundcard. Please see the bit and bit versions section for info on the difference between the bit and bit versions of n-Track. 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.
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.
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.
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. 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. You may need a splitter cable to combine the two mono inputs into a single stereo line input. When you re-record over an existing track, n-Track automatically creates a new Take for each recording attempt.
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. Disabled takes will appear in a different color purple in the screenshot above from the take that is actually being played. The Takes pop-up menu has also commands to move parts between takes, clone takes, and split a track with multiple takes into multiple tracks with one take each which is handy when you want more control over the takes selection, as for comping.
After 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, click the button on the main transport bar.
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. You can now switch between the takes by right-clicking on the track and selecting the desired take from the Takes popup sub-menu.
A popup will appear, and it will let you select both count Count-in and Punch-in modes. Selecting the Count-in menu gives the option to have a Count in before song playback [Count-in on playback] , Select the number of bars for the Count-in by selecting the [set count-in] option Note the number of bars set is in the menu option, in this case 1 measure.
You can also select if there should be a metronome click on the count-in before song playback from [Metronome on count-in]. Selecting the Punch-in menu will first display the option to enable this style of recording [Enable punch-in recording]. When you check the [Multiple takes punch-in] option, the recording process will automatically restart when the cursor reaches the punch-in end time.
This can be useful, for example, when you want to record a difficult solo and you need to try it a few times. Using this option allows you to play continuously without having to manually stop the recording, remove the bad recording from the song and start again. If the [Add takes to new tracks] option is selected, each take will end in a new track as soon as the take is finished. The 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.
Use this hold time setting to avoid very frequent stops and restarts of the recording when the signal oscillates near the threshold level.
This editing mode is called destructive because the files are actually modified. You can also use this editing mode to create loops to append the pasted part at the exact end of the current track, hold the Shift key when pasting.
To change the current editing mode, click on the editing mode icon on the toolbar which will show one of the icons above and select the desired editing mode from the drop-down menu.
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. 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. 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. 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.
Very strange and cool effects can be obtained by reversing an audio recording. 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. 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:. 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.
The dots represent the individual samples that make up the audio file. You can drag a dot vertically with the mouse to adjust the sample value. The ability to edit individual samples can be useful for correcting DC offset problems in recordings and for harmonizing editing points.
If, for example, you place two audio files next to each other and, during playback, you hear a short click corresponding to the point where the two audio files are attached, you might be able to eliminate the click by editing the samples near the connection point to make the transition visually smooth. An alternative method is to overlap the two audio files slightly and cross-fade the two. Note that abrupt changes between close samples do not usually appear in real recordings.
Manual edits of individual samples that result in abrupt changes in the sample value the screenshot above shows such an abrupt change results in very high-frequency noise bursts that can be very annoying and hurt your ears.
Normalization is the process of amplifying an audio signal so that its maximum amplitude matches a specified level. Normalization can be useful, for example, when preparing an audio file for burning a CD. Setting the maximum level of all CD tracks to 0 dB assures that no clipping occurs and that the playback level of all tracks is similar assuming that all the tracks have been processed with similar compression and limiting settings.
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 bit: converts the file to bit format Dither: Enables dithering Dither depth: Sets the depth, in bits, of the dithering noise. Noise shaping: Enables noise shaping. Selecting good cut and paste points is crucial to obtaining natural-sounding edits of audio files.
The Snap to 0 option is designed to help the selection of such points: the instances in which the waveform crosses the 0 level are often the best places at which to cut an audio file. Enable Snap to 0 in the Edit menu. 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.
Crossfading creates a smooth transition between two separate audio files. 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.
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. 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.
Saving and recalling selections is a method complementary to using markers to define edit points. 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.
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. 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. 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.
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.
Up to 25 effects can be applied to each track. 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. Select an effect from the drop-down menu. Each track signal can be sent to an aux channel using the track's send controls, located below the list of inserts effects on the mixer window.
Use send automation drawing the send evolution in the timeline to apply effects to certain parts of a track , instead of applying them to the whole track, or to vary the amount of an effect during the course of a song.
At the end of the signal path, the master channel effects process the signal resulting from the mixing of all the tracks and aux channels if needed, you exclude aux channels from the master channel effects; see aux channels settings.
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.
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. 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.
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. 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.
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.
You can apply a VST effect to a track by right-clicking on the track and selecting Effects from the pop-up menu. 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.
VST 2. The location of VST 3. Some DirectX plug-ins will refuse to work with mono tracks, reporting an error when you chose them from the list. This will also greatly enhance the result of some effects, in particular reverbs. 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.
One way to overcome this problem is to Freeze channels that are being processed by CPU intensive plug-ins. 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.
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. The Bounce command consolidates tracks made of multiple audio files into a track with a single audio file. 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.
A ReWire-compatible slave program will automatically send its signal into the corresponding n-Track ReWire channel s. 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. Closing the programs in the reverse order may cause stability problems. 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. 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.
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.
Each send can be pre-inserts, pre-fader, or post-fader. 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. 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.
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. Having to change the same setting for many tracks may be tedious; instead, send a group of tracks to a dedicated group channel. When the group channel is created, a new channel strip appears on the mixer window.
A group channel can have its own effects, can send its signal to aux channels and can have automated volume and pan just like a regular track. Group channels can in turn be sent to other group channels, allowing you to organize the song in a hierarchy of groups and allowing for great flexibility in the routing of signals. Effects can be automated using either: - Automation of effects parameters using parameter envelopes or mouse input recording.
All n-Track built-in plug-ins support parameters automation. Click on the Draw Volume Envelopes on the toolbar and choose the desired parameter track volume, pan, send or return volume etc.
The timeline window will show a line superimposed on each track. This line represents the temporal evolution of the selected parameter. 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. 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. Enable effects parameters automation : if this option is disabled, effect parameters automation will switched off. By clicking on the icon on the toolbar, the current selection will be muted; i.
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. 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. 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. 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. 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.
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 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.
Select the desired options from the box that pops up:. Each audio track has 3 EQ knobs on the mixer window. 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. 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 can enable or disable the tuner by right-clicking on the frequency response window. The signal path view is interactive. You can:. 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 bits resolution, so as to assure the minimum degradation of the sound quality due to the repeated processing even when using bit audio files, the sound quality degradation caused by a few processing steps is very hard to notice.
When using bit audio files, you may feel free to process the same file tens of times without any noticeable loss of quality. 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 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. This parameter is expressed in bits , which are referred to the current output format. 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 following data may be collected but it is not linked to your identity:.
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