Adobe audition cs6 tutorials pdf free download
This lesson initially covers using the Effects Rack, which introduces the majority of effects. The second section covers the Effects menu and discusses the remaining effects that are available only via the Effects menu. The final section describes how to work with presets, including Favorites. Using the effects rack For all lessons involving the Effects Rack, it is best to use the Default workspace. Click the Transport Play button to audition the loop, and then click the Transport Stop button.
A toolbar is located above the inserts, and meters with a second toolbar are below the inserts. You series, meaning that the audio file feeds the first effect input, the first effect output can leave empty inserts feeds the second effect input, the second effect output feeds the third effect input, between effects and and so on until the last effect output goes to your audio interface. Press the spacebar again to stop playback. Begin playback. The echoes are now in time with the music.
Keep this lesson open as you continue. Either of these actions brings the effects window to the front and opens it if it was closed. If an effect already exists in that insert, the existing effect will be pushed down to the next higher-numbered insert.
When powered back on, only effects that had been on prior to bypassing return to being on. For example, a fil- ter that emphasizes the midrange could create distortion by increasing levels above acceptable limits. To set levels, in the lower part of the Effects panel use the Input and Output level con- trols with associated meters.
These controls can reduce or increase levels as needed. Do not start playback yet. Close the Parametric EQ window. However, the massive EQ boost is overloading the output. Turn up the monitoring level enough so you can hear the distortion this causes. Reduce the Input level until meter to reset the red distortion indicators. This will likely require To reset the Input or reducing the Input to dB or so. Sometimes you want a blend of the wet and dry sounds rather than all of one or the other.
E Tip: Using the Mix 2 Drag the slider to the right to increase the amount of wet, filtered sound, and slider to blend in more drag to the left to increase the amount of dry, unprocessed sound. This is called a nondestructive process using a real-time effect, because the original file remains unaltered.
However, you may want to apply the effect to the entire file, or only a selection, so that saving the file saves the processed version. For this lesson, choose Entire File. Also note that any changes are still not amplitude and compression effects permanent until you save using either Save or Save As the file. At Amplitude and Compression effects change levels or alter dynamics.
When increasing amplitude to make a file louder, choose a low enough amount of amplification so that the file remains undistorted. With Link Sliders selected, adjusting gain for one channel changes gain equally in the other channel. Deselecting Link Sliders allows for adjusting each channel individually.
They do not go into the red so it is safe to increase the gain by this amount. The Output meter goes into the red, which shows that the gain is too high and is overloading the output. Also, try decreasing the level and listen to the results. Keep Audition open. Consider two possible applications: converting stereo to mono and reversing the left and right channels. When bypassed, the stereo image is wider. Now the signal is monaural.
Click the L tab and set operation that the the L slider to 0 and the R slider to Now the left channel consists entirely of Channel Mixer preset named All Channels signal from the right channel. Set the L slider to and the R slider to 0. Now the right conversion. When bypassed, the hi-hat is in the left channel. De-essing is a three-step process: Identify the frequencies where sibilants exist, define that range, and then set a threshold, which if exceeded by a sibilant, automatically reduces the gain within the specified range.
This makes the sibilant less prominent. Sibilants are high frequencies. Look carefully at the spectrum and confirm that you see peaks in the range around Hz. Similarly, when set to minimum Hz , the sibilants are above this range and are still audible. Adjust the Center Frequency to hear the greatest amount of sibilants and the least amount of the voice, which will be around Hz.
Dynamics processing With a standard amplifier, the relationship between the input and output is linear. A Dynamics Processor changes the relationship of the output to the input. This change is called compression when a large input signal increase produces only a small output signal increase and expansion when a small input signal increase produces a large output signal increase.
Expansion is less common; one application is to expand objectionable low-level signals like hiss to reduce their levels further.
There are also many uses for both as special effects. In the following graph, as the input signal changes from dB to dB, the output changes from dB to only dB.
As a result, the Dynamics Processor has compressed 60dB of input dynamic range into 5dB of change at the output. But from dB to 0dB, the output changes from dB to 0dB. Therefore, the Dynamics Processor has expanded 40dB of input dynamic range into 95dB of out- put dynamic range.
Choose the Default preset, which provides neither compression nor expansion. Click in the middle of segment 1 e. Drag it up a little bit to around dB.
Click on the line at dB and dB to create two more squares. Click on the one at dB, and drag it down all the way to dB. This effect makes the drums sound more percussive. Bypass the dynamics processing, and and effects. By adding Make-Up gain, the documentation for processed signal is now a little bit louder.
This is different from simple attenuation which lowers the levels of all signals , because in this example of limiting, levels below dB remain untouched. Levels above dB are compressed with an essentially infinite ratio, so any input level increase produces no output level increase above dB. This limiter also has an Input Boost parameter, which can make a signal subjec- tively louder.
In most cases the default is fine. Past a certain amount of input of thumb is to set it for the most natural sound. With no become unnatural. The level at which this will look-ahead time, the limiter has to react instantly to a transient, which is not occur varies depending possible: It has to know a transient exists before it can decide what to do with it.
With voice, instantaneous. The two most important parameters are Threshold the level above which compression starts to occur and Ratio, which sets the amount of change in the output signal for a given input signal change. For example, with a ratio, a 4dB increase in input level produces a 1dB increase at the output. With an ratio, an 8dB increase in input level produces a 1dB increase at the output.
Also, delete any currently loaded effects. This shows how the Threshold and Ratio controls interrelate, and explains why you usually need to go back and forth between these two controls to dial in the right amount of compression. Slowly increase the Ratio slider by moving it to the right. The farther you move it to the right, the more compressed the sound. Leave the Ratio slider at 10 i. The lower the Threshold, the more compressed the sound; below about dB, with a Ratio of , the sound becomes so compressed as to be unusable.
Leave the Threshold slider at dB for now. When you bypass the Single- Band Compressor, note that the meters are more animated and have more pronounced peaks. The reason is that reducing peaks allows for increasing the overall output gain without exceeding the available headroom or causing distortion.
Attack sets a delay before the compression occurs after a signal exceeds the threshold. Allowing a slight Attack time, like the default setting of 10ms, lets through percussive transients up to 10ms in duration before the compression kicks in.
Now set the Attack time to 0. There are no rules about Release time; basically, set it subjectively for the smoothest, most natural sound, which will usually lie between and ms.
You can use the same basic steps as in the previous lesson to explore the Tube-Modeled Compressor. The one obvious difference is that the Tube-Modeled Compressor has two meters: the one on the left shows the input signal level, and the one on the right shows how much the gain is being reduced to provide the specified amount of compression. It divides the frequency spectrum into four bands, each with its own compressor. Note that each band has an S Solo button, so you can hear what that band alone is doing.
This shows how multiband compression can add an element of equalization; the output gain for the two upper bands is considerably higher than the two lower bands. This is the mirror image of the Enhance Highs preset. The reason is that the highest band has an extremely low threshold of dB, so even low-level, high-frequency sounds are compressed.
Speech Volume leveler The Speech Volume Leveler incorporates three processors—leveling, compression, and gating—to even out level variations with narration, as well as reduce back- ground noise with some signals. As you move the slider to the right, the output will become louder than the input. Choose a value of about 60 for now. The output will peak at around -6dB. Adjust the Target Volume Level until the peaks match the peaks you saw in step 6. The slider should be around dB.
There should be fewer volume variations between the soft and loud sections. To best hear how this works, with the Leveling Amount at the default setting of , select the audio between 2 and 6 seconds, and loop it. Move the Leveling Amount back to 60, and the noise goes away. Observe the meters, and see if further tweaking can help create a more consistent output. In the reducing the attack and illustration the top waveform is the original file, whereas the lower one has been decay time of the effect processed by the Speech Volume Leveler.
Delay and echo effects Adobe Audition has three echo effects with different capabilities. All delay effects store audio in memory and then play it back later. The time that elapses between storing it and playing it back is the delay time.
This makes it easy to hear single delay. Leave Adobe Audition where the project has a particular tempo. Samples is useful for tuning out short timing differences, because analog Delay you can specify delays Before digital technology, delay used tape or analog delay chip technology.
These down to 1 sample at a 44,Hz sample produced a more gritty, colored sound compared to digital delay. Analog Delay simply repeats the audio with the start time of the repeat specified by the delay amount. Unlike the Delay effect, there are separate controls for Dry and Wet levels instead of a single Mix control.
The Delay slider provides the same function as the Delay effect except that the maximum delay time is 8 seconds. No Feedback a setting of 0 produces a effect. For 4 With feedback at 50, set the Trash control to Vary the loop tempo is feedback, being careful to avoid excessive, runaway feedback.
Keep Audition open for the note is The Lesson04 folder includes a file called Period vs. For example, if the response is set to be brighter than normal, each echo will be brighter than the previous one. E Tip: To set both 2 Compared to the previous delay effects, Echo has yet another way of setting the channels to the same echo mix; each channel has an Echo Level control that dials in the echo amount. Delay Time, enable Lock Left and Right.
The Dry signal is fixed. That makes it easier to hear the difference moving a single slider has on the sound. Now the echoes are brighter. The echoes are now bassy. Filter and eQ effects Equalization is an extremely important effect for adjusting tonality. Adobe Audition has four different equalizer effects, each used for different purposes, that can adjust tonality and solve frequency-response related problems: Parametric Equalizer, Graphic Equalizer, FFT Fast Fourier Transform Filter, and Notch Filter.
Parametric equalizer The Parametric Equalizer offers nine stages of equalization. Each parametric equalization stage has three parameters. The Parametric Equalizer is capable of high amounts of gain at the selected frequencies.
Start playback. Each represents a controllable parametric stage. Click one of them e. Drag left to affect lower frequencies or right to affect higher frequencies. Listen to how this changes the sound. The L and H squares control a low shelf and high shelf response, respectively. This starts boosting or cutting at the selected frequency, but the boost or cut extends outward toward the extremes of the audio spectrum.
Note how this increases the treble. Similarly, click on the L box to hear how this affects the low frequencies. There are two additional stages, Highpass and Lowpass, which you enable by clicking on the HP and LP buttons, respectively.
Click those buttons now. A Highpass filter is helpful for removing subsonic very low-frequency energy. Click on the HP box and drag it to the right to hear how it affects low frequencies. Note how this creates a gradual curve. Keep this project open for the next lesson. The screen shot shows a steep along the bottom of Highpass slope, a slight parametric boost with stage 2, a narrow parametric cut the screen has three additional options.
Constant Q, where Q is a ratio compared to frequency, is most common, whereas Constant Width means the Q is the same regardless of frequency. The Ultra-Quiet option reduces noise and artifacts but requires much more processing power and can usually be left off. Range sets the maximum amount of boost or cut to 30dB or 96dB. The more common option is 30dB. Caution: In the following lesson, keep monitor levels down as you make adjustments.
The Graphic Equalizer can produce high amounts of gain at specific frequencies. Move the various sliders up and down to hear how each affects the timbre through varying the level within their respective frequency bands. In musical terms, each slider is an octave apart.
Keep Audition open in preparation for the next lesson. P Note: The strip along the bottom of the Graphic Equalizer screen has three additional parameters.
Range sets the maximum available amount of boost or cut up to dB which is a lot! Accuracy affects low-frequency processing. Otherwise, leave it at the default of points to reduce CPU loading. Master gain compensates for Output level changes caused by using the EQ. Turn it down if you added lots of boosting; turn it up if you used lots of cutting.
To hear how it works, follow the same basic procedure as the lesson for the 10 Bands version. The default settings are a practical point of departure. FFT is a highly efficient algorithm commonly used for frequency analysis. You can then drag this point up, down, or sideways. You are not limited to the number of points you can add, which allows you to make very complex—and even truly bizarre—EQ curves and shapes. The screen shot on the left shows Spline Curves deselected and the original placement of points, whereas the screen shot on the right shows Spline Curves selected.
P Note: As for other FFT Filter parameters, for Scale choose Logarithmic when working primarily with low frequencies because this produces the best resolution for drawing in nodes.
Linear has the same advantage at high frequencies. For the Advanced options, for the best accuracy with steep, precise filters, choose higher values like to Lower values produce fewer transients with percussive sounds. For Window, Hamming and Blackman are the best overall choices.
The choices listed first narrow the shape of the response curve with subsequent choices progressively widening the shape. Note the huge amount of hum in the file. Turn off notches 3, 4, 5, and 6. Turn off notches 1 and 2. Experiment with the Gain parameters for notches 1 and 2. These tend to produce very specific sounds, and the presets included with Adobe Audition are a good place to start.
But there will also be some analysis of which parameters are most important for editing. The Chorus effect is optimized for stereo signals, so convert mono signals to stereo for best results. Then click OK. Play the file to hear what it sounds like. Select Highest Quality; most modern computers can provide the additional processing power this option needs. If the audio crackles or breaks up, deselect this option.
Notice how the sound becomes more animated. To make this more obvious, increase the Modulation Rate to 2. Return Modulation Depth to 0. Because this adds a lot more audio, you may need to bring down the Output control in the Effects Rack panel to avoid distortion.
Set it to around 40ms for now. Set it to around ms. Stereo Field makes the output narrower or wider. If you like the sound better, leave them selected. Note that some of the more bizarre sounds combine lots of modulation, feedback, or long delay times. Alter the Feedback setting; more feedback produces a more resonant sound.
Stereo Phasing changes the phase relationship of the modulation; when set to 0, the modulation is the same in both channels. Increase the Phasing amount to offset the modulation in the two channels, which creates more of a stereo effect. Vary the Modulation Rate to change the modulation speed. Experiment with these options. Selecting Inverted changes the tone. The effect varies depending on the other parameter settings. Many of the more radical patches use either high Modulation Rates, large amounts of Feedback, longer Initial or Final Delay Times, or a combination of these.
Speed provides the same function as Modulation Rate. Phaser The Phaser effect is similar to Flanging but has a different, and often more subtle, character because it uses a specific type of filtering called an allpass filter to accom- plish its effect instead of delays. Play the file.
Change the Upper Freq to around Hz. The farther you move the Phase Difference away from the center 0 position, the greater the stereo effect. Leave it at for now. Leave it at 0. Note how at faster settings the effect is almost like vibrato. Return it to 0. This complements the Upper Frequency parameter, which is the highest frequency that Modulation attains. Moving the value toward 0 increases the proportion of dry signal to wet signal, whereas moving the value toward increases the proportion of wet signal to dry signal.
Experiment with these parameters to hear how they affect the sound. These include the ability to remove noise, delete pops and clicks, minimize the sound caused by scratches in vinyl records, reduce tape hiss, and more. Two common reverb processes are convolution reverb and algorithmic reverb. Audition includes both. Convolution Reverb is generally the more realistic sounding of the two.
It loads an impulse, which is an audio signal typically WAV file format that embodies the characteristics of a particular, fixed acoustic space.
The effect then performs convolution, a mathematical operation that operates on two functions the impulse and the audio to create a third function that combines the impulse and the audio, thus impressing the qualities of the acoustic space onto the audio.
The trade-off for realism is a lack of flexibility. Algorithmic Reverb creates an algorithm mathematical model of a space with variables that allow for changing the nature of that space. All Audition reverbs other than the Convolution Reverb use algorithmic reverb technology. Each type of reverb is useful.
However, it is a CPU-intensive process. Note how each impulse produces a different reverb character. Move the Damping E Tip: You can use LF slider to the left to simulate the effect of a room with lots of sound-absorbing Convolution Reverb to load most WAV files material, which absorbs high frequencies more readily than low frequencies. Online sources offer free impulses that work with 8 Pre-Delay sets the time before a sound first occurs and when it reflects off a standard convolution surface.
Also, you can load phrases, loops, slider to the left to narrow the image. These can be valuable for sound design and Studio reverb special effects. Many of the Full Reverb and Reverb parameters cannot be adjusted during playback, because they are very CPU-intensive. Drag the minimum, can add a Decay slider all the way to the left, and then vary the Early Reflections slider. Increasing early reflections creates an effect somewhat like a small acoustic This can make narration space with hard surfaces.
Adjust the Width control to set the stereo imaging, from narrow 0 to wide Move the slider more to the left to reduce the high frequencies for a darker sound or more to the right for a brighter sound. The difference between damping and High Frequency Cut is that damping applies progressively more high-frequency attenuation the longer a sound decays, whereas the high frequency cut is constant.
Experiment with damping. In general, high-diffusion settings are common with percussive sounds; low-diffusion settings are used with sustaining sounds e. Also, you cannot adjust the reverb characteristics in real time—only when playback is stopped. You can edit the dry and wet levels at any time.
Leave Audition open. Full reverb Full Reverb is a convolution-based reverb and is the most sophisticated of the various reverbs but also the most impractical to use because of the heavy CPU loading. No parameters other than the level controls for dry, reverb, and early reflections levels can be adjusted during playback, and even then, the level control settings take several seconds to take effect however, if you stop playback and adjust them, the change occurs immediately on playback.
Also, if you change any of the non-level reverb parameters while stopped, it can take several seconds before playback begins. Stop playback. Now vary the Noise Floor slider for the best compromise between hiss reduction and high-frequency response; -2dB is a good choice. Reducing clicks Clicks can consist of the little ticks and pops you hear with vinyl recordings, occa- sional digital clocking errors in digital audio signals, a bad physical audio connec- tion, and so on.
Conversely, a setting of lets through too many clicks. Choose a setting of 20 to reduce most clicks while minimally affecting the audio. Higher settings allow Audition to recognize more complex clicks but requires more computation and may degrade the audio somewhat.
This is not a real-time control, so you need to adjust it, play the audio, adjust, play, and so on. For now, click the Transport Stop button, and then move the slider to Because this is a computation-intensive process, during real-time playback Audition may not be able to process a click prior to playing it back. Note that this closes the Automatic Click Remover processor.
Move the complexity slider to 35, and click Apply again. Click the variations. If these sounds are relatively constant, Audition can reduce or remove them using the Noise Reduction process. This process can also reduce hiss and allow for more detailed editing compared to the Hiss Reduction option. As with reducing hiss, Audition will take a noise print of the hum and subtract only this objectionable noise from the file.
The hum will be gone during silent sections. A setting between 10 and 20dB is a good compromise between affecting the audio and reducing the noise. Set this slider to 15dB for now. Click the Advanced disclosure triangle for more options.
Close Audition without saving anything click No to All in preparation for the next lesson. Removing artifacts Sometimes particular sounds will need to be removed, like a cough in the middle of a live performance. Audition can do this using the Spectral Frequency Display, which allows for editing based on not just amplitude and time as with the standard Waveform Editor , but also frequency.
This exercise shows you how to remove a cough in a perfor- mance by classical harpsichordist Kathleen McIntosh. A circular cursor that looks somewhat like a bandage icon appears. Adjust the size so that the circular will vary depending on the Spectral Frequency cursor is as wide as the cough. Be careful to drag over only the cough.
The healing process takes audio on either side of the deleted audio, can try multiple times, and through a complex process of copying and crossfading, fills in the gap or even use this process caused by removing the artifact. Alternate click removal You can use the Spectral Frequency Display to remove clicks. Although this is a manual process that is more time-consuming than using the Automatic Click Remover effect, the removal process will be more accurate and have less impact on the audio quality.
This selects the noise in both channels. The file sounds as if the clicks had never been there. This lesson takes a drum loop and uses the Spectral Frequency Display to remove four drum hits. Move both Gain sliders full left, and click Apply. With both Gain sliders full left, click Apply. Most of the sound from the hits is gone, but you can still hear a little bit of noise on the second and fourth hits.
The drum loop is the same as the original but without the four hits. P Note: The technique of drawing a lasso around artifacts can remove sounds like finger squeaks on guitar strings, clicks or pops, breathing while a person plays an instrument, and many other artifacts. However, with sufficient practice, this type of restoration is extremely effective. Name two ways other than restoration where these tools can be useful. Review answers 1 Automatic click removal is faster, but manual click removal can be more effective.
Subtracting this from the audio file removes the noise. As the final link in the music production chain, mastering can make or break a project. As a result, people often hand off projects to veteran mastering engineers, not just for their technical expertise, but to enlist a fresh, objective set of ears. However, if your goal with mastering is simply to make a good mix better, Audition provides the tools required for professional-level mastering.
The more you work with mastering, the more your skills will improve. In addition, remember that ideally the purpose of mastering is not to salvage a recording, but to enhance an already superb mix.
Note how this tightens up the low end. The difference is subtle, but often mastering is about the cumulative effect of multiple subtle changes. Sweep the Frequency back and forth between 20 and Hz, and note that the kick really stands out around 45Hz. Note that in the screen shot, unused bands have been turned off for clarity. Keep Audition open for the next lesson. This can also help the music overcome background noises found in many different listening environ- ments.
This lesson uses the Multiband Compressor to control dynamics. Adding effects afterward could more apparent loudness. Now drag up to 0. Also note that the Multiband Dynamics has emphasized being applied. After choosing the preset, click the Transport more cohesive feeling. Play button. Stereo imaging stretches the stereo image so the left channel moves more to the left, and the right channel moves more to the right.
In this lesson, increasing the stereo image helps separate the two guitars in the opposite channels even more. Only the Widener will be used. P Note: The Apply function resets all effects to their default value. P Note: It can take years to become good at the art and science of mastering, especially if the file has problems 2 Play the file to hear the result of boosting these four drum hits.
EQ, dynamics, and some selected other 4 Click Apply. Audition applies the result of all the effects to the file and removes processors. Normally ambience is added during the mixing process but can sometimes improve the sound when added while mastering. Sound design can refer to music, but this lesson emphasizes sound effects and ambience.
These types of sounds are also common in sonic logos like the sound you hear to identify Intel Inside or NBC or the sounds layered in movie scenes to create a particular mood.
Occasionally, against this engine sound backdrop, sudden, sharp sounds might appear to indicate a new leak breaking through the engine wall. Sound effects libraries are available from several companies, but a sound designer will often modify these or record sounds using a field recorder. For example, in the beginning of the movie Raiders of the Lost Ark, a giant boulder rolls toward Indiana Jones. The sound of the rolling boulder was created by taping a microphone to the rear bumper of a Honda and recording the sound of the car backing down a gravel driveway.
Subsequent sound design work turned this into a huge, ominous sound. Two files were recorded for the following lessons using a portable digital recorder: water running into a sink from a faucet and a wall fan.
Creating rain sounds With sound design, it helps to start with a sound in the same genre. To create rain, the running water would most likely produce a better end result than the recording of the fan. Click the Transport Play button to audition the loop. Ensure that Range is 48dB and Master Gain is 0. With the EQ power on, you now have a light spring rain.
You might find the effect more realistic if you bring the 8k slider down to dB. In this case there would be fewer high frequencies due to the house walls and windows blocking the highs. Keep this project open as you move to the next lesson. Set and to 0, to , 1k to , 2k to , and 4k to As the character moves closer to the brook, increase 2k to and 4k to Now the brook sounds closer.
Keep this project open as you move on to the next lesson. This lesson demonstrates how signal processing can turn one sound into something completely different. It will take several seconds for processing to occur. Click the Transport Play button pitch shifting can add anomalies, like volume to hear the processed sound.
If so, make an audio reverb. Stop playback by clicking the Transport Stop button so the first and last two you can select a reverb preset. Extreme shifting may also lower the volume, 8 Click the Transport Play button, and listen to your refined insects at night sound. The settings should be the same as they were previously, but if not, choose the preset Default from the drop-down menu.
Normalize All Channels Equally should also be selected. The Reverb should still have the Great Hall preset selected. Move the Decay, Width, Diffusion, and Wet sliders all the way to the right. Move the Dry slider all the way to the left. When you click the Transport Play button, you should now hear an ethereal, animated pad. Select the preset Spooky. Select the preset 10 Voices. Note how extensive pitch stretching coupled with effects from the Studio Reverb, Echo, and Chorus can turn running water into an alien soundscape.
Creating sci-fi machine effects Just as you used running water to generate water-based effects, the fan sound makes a good basis for machine and mechanical sounds. This lesson describes how to turn an ordinary wall fan sound into a variety of science-fiction, spaceship sound effects. Click the Transport Play button to hear what the file sounds like.
Click OK; processing will take a few seconds. From the Presets drop-down menu, choose Drum Suite. Click the Transport Play button, and note how the sound becomes more metallic and machine-like. If the preset Default is not already selected, choose it from the Presets drop-down menu. Pull the H box all the way down and left to around 3kHz.
Click the Band 2 button to enable that parametric stage. Set the Decay, Width, Diffusion, and Wet sliders all the way to the right. Click the Transport Play button. Leave this project open in preparation for the next lesson. Creating an alien drone flyby In addition to creating static sound effects, Audition includes a Doppler Shifter processor that imparts motion—from left to right, around in circles, tracing an arc, and so on.
Choose the preset Lowest Fidelity from the Presets drop-down menu. Click the Transport Play button to hear the drone fly by from left to right. Using the slider would make it almost impossible to choose this precise a value. This one of these lessons. If necessary, review Lesson 1 to make sure you understand the principles behind interfacing.
This lesson assumes you know how to map inputs and outputs. The input level should never interface will have either a physical meter or dedicated control panel software that go into the red on a level meter. Enter the File Name RecordNarration. However, for this lesson choose Hz—the standard for CD audio—as the sample rate from the drop-down menu. Sample rates lower than Hz are of very low fidelity and are generally used for speech, dictation, toys, and so on.
Most studies show that few people can accurately tell the difference between and or higher sample rates. Higher sample rates also require more storage space; that is, a one- minute 96kHz file will require twice the space as the same file at 48kHz.
For a mic or electric guitar, choose Mono. For a portable music player or other stereo signal source, choose Stereo. The option 5. Files that are bits take up 50 percent more space than bit files, but given the low cost of hard disk storage, this is an acceptable trade-off.
They take up more space, and many programs are not compatible with bit float files. For Windows users, also verify that the Device Class and Device settings are correct. Dock the meters horizontally along the bottom of your workspace for the best meter resolution.
Recording begins immediately. Speak into the mic or play back whatever sound source connects to your interface. Recording will pause, but the meters will still show the incoming signal level. The waveform you recorded will be selected.
If you 16 Record for a few seconds, and click Stop. The audio between where you started make a mistake in a particular line, click at and stopped recording will be selected. This lesson shows you how to record into practice to have a separate, high-speed the Multitrack Editor, explains how to transfer your recording into the Waveform at least rpm Editor for editing, and then describes how to return it to the Multitrack Editor. This separates A dialog box appears with fields for an editable File Name, Folder Location for streaming audio on the audio drive from storing the project, and project Template.
The dialog box also includes Sample program operations on Rate and Bit Depth fields identical to those in the New Audio File dialog box, your main drive. Also, and a Master field that determines whether the output is Mono, Stereo, or a backing up the audio drive backs up all of surround format if supported by your interface.
For this tutorial, accept the default folder. If you want to edit these settings, select None for Template. The Multitrack Editor opens. A track can be stereo or mono. Record your signal source for at least 15—20 seconds. As when recording in the Waveform Editor, you can pause and resume recording by clicking the Pause button.
Then click Stop. Notice that there is a separate layer recorded on top of the previously recorded track. By clicking on it, you can drag it left or right, as well as drag it down into Track 2. P Note: To transfer the file back into the Multitrack Editor at the position from which it came, click the Multitrack button or type 0.
The file can be in any file format that Audition recognizes; in the Multitrack Editor, it will automatically be converted to the project settings. For example, if you created a project with bit resolution, you can bring it in as a bit WAV file or even an MP3 format file. With the Waveform Editor, you simply drag the file from the desktop into the edi- tor.
This lesson demonstrates how to drag and drop a file into the Multitrack Editor. Leave the default folder location as is, and select 24 Track Music Session as the template.
A yellow line appears where the dragged file will start; this should be flush with the beginning of the track. Release the mouse button. You can click on this line and drag up or down 6 These tracks were recorded at a high volume, and they are overloading the to decrease or increase, output. The sound is now undistorted, and none of the meters go into the red clipping zone. However, you can access a func- tion that extracts audio from CDs and places it into the Waveform Editor. You can then edit it or transfer it to the Multitrack Editor as described previously.
You need a standard audio CD to complete this lesson. A dialog box appears that shows the drive, and if the computer is connected to the Internet, Audition retrieves information from the freedb database to populate the Track, Title, and Duration fields.
Leave it at the default Maximum Speed ; if errors occur during the extraction process, choose a slower speed. They will be extracted one at a time, each as a separate file. You need to extract only one track for this lesson, so click the Toggle All button to deselect all tracks, and then select one track.
The extraction process begins and places the audio in the Waveform Editor. Audition includes several template files for Multitrack Sessions, but you can also create your own. These settings will be incorporated into the template. Click Session Templates, click Open, and then click Save. Click Session Templates, and then click Save. Next time you open a new Multitrack Session, the template you created will be available in the drop-down menu of templates.
In the Waveform Editor, a single clip is the only audio element. A multitrack pro- duction assembles multiple audio clips to create a musical composition. For example, one track could contain drum sounds, another bass, a third vocals, and so on. A track can contain a single long clip or multiple short clips that can be identical or different. A clip can even be positioned on top of another clip in a track however, only the clip that is on top will play back , or clips can overlap to create crossfades described in Lesson This involves recording or importing audio into the Multitrack Session.
For example, with a rock band, tracking could consist of recording drums, bass, guitar, and vocals. These might be recorded individually each player records a track, typically while listening to a metronome for reference in particular combinations e. This is the process of recording additional tracks. For example, a singer might sing a harmony line to supplement the original vocal.
After recording the tracks, editing can polish them. For example, with a vocal track you could remove the audio between verses and choruses to reduce any residual noise or leakage from other instruments. You might even alter the arrangement, like cut a solo section to half its original length. After editing, the tracks are blended together into a final stereo or processes are available surround file. The mixing process primarily involves adjusting levels and adding only in one editor or the other.
If a menu effects. A Multitrack Session in Audition CS6 provides the tools to do basic option is gray in either editing tasks during the mixing process, but if detailed edits are needed, Session editor, that option is not audio can be transferred to the Waveform Editor for further editing. Multitrack and Waveform Editor integration A unique Adobe Audition feature is that it offers different environments for wave- form editing and multitrack production.
In addition, these are not isolated from each other. This lesson illustrates how the Multitrack and Waveform Editors work together. Like the Waveform Editor, the Multitrack Editor plays linearly from However, there is only one physical clip, which start to finish. However, because the Multitrack Editor consists of multiple, is stored in RAM; the parallel tracks, each of which can play back a clip, multiple clips can play back graphic clips in the simultaneously.
Play the song to become familiar with it. Multitrack Editor just reference the physical 3 To check out the Waveform and Multitrack Editor integration, click the first clip clip. It shows all the clips times, as instructed by in the Multitrack Session so that you can choose any of them for editing. The looped area will play continuously. One section has a fixed set of controls, whereas the other section is an simultaneously by placing the cursor over area whose controls change according to a particular selected function.
To reveal the track controls area these controls, complete the following steps. Main track controls P Note: If both the The main track controls are the most commonly adjusted parameters for mixing. Note that the Mute button is green when active and gray when it is off not muted. Only the Main Drums track will sound. The R button is for recording; do not click it for now. Leave the Main Drums track soloed for all the following lessons unless instructed to do otherwise.
P Note: There are two solo modes—Exclusive soloing one track mutes all other tracks and Non-Exclusive the default; you can solo multiple tracks simultaneously. To override either mode, Ctrl-click Command-click on the Solo button. On the same row as the Mute and Solo buttons, red lights will remain click the waveform icon to the left of the Percussion track, and drag it up until a lit so you know the yellow line appears at the bottom of the Main Drums track.
To reset the red lights and turn them off, click on the lights. P Note: If dragging the Pan knob left places in the audio to the right, check the connections going to your speakers and audio interface.
For now, leave the Volume at 0. This changes the track position in the stereo field. Drag right, and the audio will play from only the right speaker or headphone. Click on it again to return to stereo. As a result, each track has the option to insert a Parametric EQ effect.
Note that the EQ area now shows the EQ curve. This lesson covers using the features that differ. Increase the track height so you see 16 slots—just like the Effects Rack in the Waveform Editor. Click the Transport Play button; you should hear lots of repeating echoes. Click it to turn off the Analog Delay effect, and then click it again to turn the Analog Delay effect back on. It turns red to indicate that the effect is now post-fader. Although buses appear in the Multitrack Editor like tracks and have several elements in common, they serve a different purpose.
A bus does not contain audio clips but instead carries a specific mix of one or more tracks. Every Multitrack Session has at least one bus—the Master bus, which provides the Master Track output. This can be mono, stereo, or 5. This is essential because as you add more tracks to a composition, the output level increases. Eventually, it will likely start distorting, but you can use the Master Track Volume control to adjust the output level and prevent distortion.
A common example is reverb, which creates the illusion of those tracks playing in a common acoustical space, like a concert hall. With older computers or highly complex projects, using more effects can slow down performance and possibly even reduce the total track count. For example, the bass player will want to lock the rhythm to the drums and will likely want to hear louder drums compared to the other instruments.
However, the vocalist, who is paying more attention to the melody, will likely want to hear more melodic instruments, like piano and guitar. The bus going to the vocalist would have more send from the piano and guitar, whereas the bus going to the bass player would have more send from the drums. Note that the Master Track Volume too much. If it becomes control is set to At no point does the much more than dB Master Track output meter go into the red.
This allows 5 Alt-click Option-click the Master Track Volume control to return it to zero, the Master bus Volume and then click Play to start playback. Return the Master Track Volume control to Each track has a Sends area. You can create buses in this area, as well as control bus levels and other parameters. This creates a Bus immediately right-clicking Control- clicking in a blank below the Main Drums track. The bus appears immediately below the track where you right-clicked.
Also, note that when you create a bus, its output is assigned automatically to the Master bus as the default. Note that the bus name changes automatically in the Main Drums Send area drop-down menu.
Because you created a Reverb bus, it appears in the list of available send destinations. Choose Reverb bus. Start by clicking the fx button in the track toolbar. The bus Volume control sets 12 Click the Transport Play button to begin playback. Adobe Illustrator CS5 Essentials. Size : Adobe Captivate 8. Adobe Spark Getting Started. Adobe Dreamweaver Essentials. Size : 2 MB Downloads : Adobe photoshop tutorial.
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This lesson takes a drum loop and uses the Spectral Frequency Display to remove four drum hits. Online sources offer free impulses that work with 8 Pre-Delay sets the time before a sound first occurs and when it reflects off a standard convolution surface. The Waveform Editor includes zoom but- tons, but you can also open a Zoom panel with these same eight buttons if you want to position them elsewhere or float them. This exercise shows you how to remove a cough in a perfor- mance by classical harpsichordist Kathleen McIntosh. Four files are selected. Adobe Captivate 9 – Accessibility. The screen shot shows a steep along the bottom of Highpass slope, a slight parametric boost with stage 2, a narrow parametric cut the screen has three additional options. For most Audio Hardware offers Macs, the choices are , , , and kHz; choose , which additional level controls, is the standard for CDs. Click the variations.
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Subsequent sound design work turned this into a huge, ominous sound. Then drag down to about E Tip: Using the Mix 2 Drag the slider to the right to increase the amount of wet, filtered sound, and slider to blend in more drag to the left to increase the amount of dry, unprocessed sound. Laptops may also have an internal microphone you can use; however, using a line-level device is recommended, and the lessons will reference that type of input. The screen shot on the left shows Spline Curves deselected and the original placement of points, whereas the screen shot on the right shows Spline Curves selected. Audition algorithm is less CPU-intensive, but 8 The Advanced parameters are mostly important when manipulating voice; make the Radius algorithm sure the Solo Instrument or Voice and Preserve Speech Characteristics check has much better fidelity, and most modern boxes are selected, and adjust the Pitch Coherence slider for the best sound computers will have no quality this will be subtle. Leave the Threshold slider at dB for now.