Digital Mastering
Most Mastering today is done digitally, therefore it is referred to as
Digital Mastering. This simply means that the recording is done using computers and software instead of analog gear. Audio can however receive
Digital Mastering and still be ran through some analog gear, and visa versa.
Digital Mastering tends to be much easier to work with and understand. One of the reasons for this is the fact that computers allow you to see what you are doing. Many software programs give you the opportunity to see what you may not hear. This is a great advantage in an imperfect listening environment. And no matter how good your room is, it is never going to be perfect.
In the digital domain the peak level may deviate from the peak level in the analog domain.
There are two main reasons for this:
1. Basic sampling theory. Sampling occurs at regular intervals, and at frequencies near integer fractions of
fs, such as fs/4 and fs/2, the phase of the signal compared to the sampling times may generate a digital
peak value somewhat below the analog peak value. If the signal is not exactly at one of the critical
frequencies mentioned above, the peak value in the digital domain will get very close to the analog peak
value. If the analog signal prior to sampling was properly bandwidth-limited, the output after digital to
analog conversion will be substantially equal to the analog input signal.
2. Gibb's phenomenon. Occurs when limiting the bandwidth of a wide-band signal (or truncating an
impulse response). This is particularly important when the signal is clipped in the digital domain, but it
applies generally. What happens is that a square wave (or hard clipped signal) can be viewed upon as a
sum of individual sine waves of frequencies 1, 3, 5,... times the fundamental frequency. The flat top of the
square wave depends on the presence of all harmonics at the right levels and phases. If some of the
harmonics are removed by lowpass filtering, the peak value of the signal rises. When converting from
digital to analog a low pass filter is always applied, so the analog level may be higher than expected.
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