LAB SIX IN AMADEUS

RECORDING AUDIO: AMADEUS

Recording sounds into the computer can be done using an audio
editor or ProTools. Audio editors are designed to record sounds, and
the process is therefore more straightforward in these programs.
Before you can record in a program like Amadeus, you will
need to make sure certain things are set up correctly. This section
assumes that you are using a microphone connected to a mixer, and
the mixer is connected to the Line In of the computer. If this is not
the case (for example, your microphone is connected directly to an
Mbox), you cannot record into Amadeus, and must use ProTools.

SETTING THE SOUND PREFERENCES

First, make sure that the computer is set to record a signal from the
microphone (rather than the internal CD Player).

Go to System Preferences, found under the Apple menu.

Click on the Sound icon at the end of the second row.

Select Input in the upper tabs. Make sure that Line In is the
selected input device. This will force Amadeus to look for
input from the Line input on the Mac.

The Sound Preferences dialog box

Selecting Internal microphone (if available) will choose the
microphone that is built into the Macintosh itself, which is
invariably of poor quality.

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GETTING READY TO RECORD

Launch Amadeus.
Click on the Record button in the Navigator, or select Record
from the Sound menu:

Getting ready to record

The Record Sound dialog box will be presented. Select the
Input tab:

The Record Sound dialog box in Amadeus

Make sure the Driver Type is set to CoreAudio (which means
the settings you made in the Sound Preferences will be used,) and
the Driver is Built-in Audio. Lastly, the Source should be set to
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Line In in order to record an audio signal connected to the

computer (via a mixer).
Playthrough sends the incoming signal (from the microphone)
directly to the output of the computer. This allows you to hear the
input source through your monitors. In most cases, this is
advantageous; however, it has the potential for feedback. Even if
the feedback is slight (for example, the microphone picks up some
of the sound coming through the speakers, but not to the extent
that it causes amplification and uncontrollable screeching), this will
result in a alteration to the sound caused by phasing. For this
reason, once you can hear the signal through your monitors, turn
down the level on them, or turn off Playthrough.
Click on the Record tab. Check your signal level by playing
your sound into your microphone

The Record Sound dialog box, adjusting the input level

Test your sound by playing it; you should see the VU meter
display something. Adjust the Gain (1) slider until the Level VU
displays a good signal without going into the red.

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Click OK when you are ready to record (recording begins

right when you click on the OK button).

RECORDING AN
EXTERNAL SOURCE
Once you click OK, the timer will begin counting, displaying
the length of time you have been recording:

The timer in Amadeus

While you are recording, watch the VU meters to make sure
that you are not distorting.
Click the OK button when you have finished recording.
The newly recorded audio file will then appear in a new
window, entitled Untitled Sound.

IF YOUR LEVELS ARE NOT IDEAL

A major difference between analogue and digital recording is the
ability to see your recorded levels immediately; you can instantly
tell if your recording is distorted or has very low levels.
In the case of distortion, once the input signal has been clipped,
there is nothing that can be done to correct the problem; you will
have to record your source again either by using lower levels on the
mixer or by moving the source further away from the microphone.
In the case of low levels, it depends on how low they in fact
are. Remember, the signal-to-noise ratio is much higher on digital
recordings (96 dB), and there is no background noise being added.
Therefore, when you view your audio file, it isnt necessary for the
maximum amplitude to be close to the edges after you record.
For example, the audio file below is a good recording:

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Viewing the amplitude of a soundfile

Notice that the maximum displayed amplitude doesnt appear
to be much higher than 50 per cent. However, a visual gauge of the
amplitude is not necessarily the best indication of the recording
levels. Remember that there are thousands of amplitudes (samples)
taken per second; in the above example, the five seconds of audio
contain over 200,000 samples. The display on the computer can
only represent a limited number of samples, based upon the
resolution of your monitor. For example, even a setting of 1280 by
854 pixels will, at most, display 1280 of the 200,000+ samplesone
pixel per 156 samples. Furthermore, a spike or transient, which
amounts to a sudden jump in amplitude for a short duration
(perhaps only a hundred samples), may not be represented at all.
Therefore, viewing the entire audio file can give you a rough
estimation of the relative amplitudes but not an accurate measure
of the maximum amplitude.
For this reason, Amadeus and many other audio editors will
calculate the actual maximum amplitude for the file. In Amadeus, this
is done via the Analysis menu, Waveform Statistics. Select the
entire file, and then select this menu item.
This brings up a lot of complex information:

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Sample data from a file, in Amadeus

Of most concern to us is the Peak Amplitude (-2.297 db in this
case), and the number of cliiped samples (0 in this case, which is
good).
In the example above, the maximum is 2.297 decibels below the
maximum, which amounts to about seventy-nine per cent of the
maximum possible. This represents a single sample value of 26,053
out of a possible 32,768, which is half of 65,536 (half, since negative
numbers take up the other half), or sixteen bits.
But how can the amplitude be 2.297 dB lower than, but also
seventy-nine per cent of, the maximum? Isnt that something of a
contradiction?
Remember that the decibel scale is a nonlinear logarithmic
measurement because our perception of amplitude is nonlinear.
Doubling the amplitude of a sound will not create twice the
decibels; instead, the new sound will only be 6 dB louder.
Therefore, a sound that is half as loud as the maximum (unity gain)
will be 6 dB lower.
In our example, seventy-nine per cent seems far from ideal
only a B on the grade scale! However, attempting to rerecord the
signal with higher levels may prove fruitless since you will
probably begin to distort the signal (if it goes above 0 dB). For this
reason, we can amplify the signal to a maximum level digitally.

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NORMALIZING AN
AUDIO FILE IN AMADEUS
With an audio file open, select Normalize from the Effects menu.
Doing so will bring up the Normalize dialogue box, which
allows you to specify the exact ratio of normalization in terms of
decibels relative to the maximum (0 db being a the maximum
without distortion) or a maximum percentage. Since normalizing
can never distort the signal, setting it to 0 db, or 100 per cent is safe.
(Some audio engineers will set the normalization percentage
at about ninety-six per cent because they feel older CD players
cannot handle the absolute maximum digital range.)

Amadeuss Normalize dialogue box.

Once you have set the percentage, click OK to normalize your
audio file.

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