Did you know about 0 dBFS+? I did not until some time ago, when I came across TC Electronics’ site, where several
papers about pro audio are presented. One of them is about
0 dBFS+. Other interesting articles exist on related topics,
Dealing With Peaksby
by Bob Katz and
Taming Wild Mastering Levels by
David Moulton. They require more than one reading in order to fully grasp the concepts, but I think they deserve the time needed to read and study them. At least for people interested in serious digital audio.
So, what is 0 dBFS+? It refers to the 0 dBFS reference level (FS stands for Full Scale) in digital audio, that is the maximum allowed level for the digital representation of signals. The term 0 dBFS+ means that what appears to be a perfectly “legal” digital signal (a “full scale” or 0 dBFS signal, with no “digital over”) can cause distortion when it is converted back to the analog form (during D/A conversion). What is encoded at full scale in the digital domain can actually be the representation of an analog signal larger than full scale. This can overload the D/A converter, if it is (poorly) designed thinking that signals larger then “full range” will never turn out, and therefore cause distortion.
To illustrate the concept you should be familiar with the sampling (Nyquist) theorem, which is the foundation of digital signal processing (DSP). With a sampling frequency of 44100 Hz, take a 11025 Hz sine wave with a 45 degree phase and sample it. Imagine that the A/D converter range is -1 Volt to +1 Volt, and the sine wave amplitude is exactly 1 Volt. The actual samples are +0.707, +0.707, -0.707, -0.707 in amplitude. Now take a “simple” normalization program. It thinks that the signal is below digital full range, so it amplifies the samples 1.414 times (that is, +3 dB), and they become +1, +1, -1, -1. This is a fully “legal” digital signal, but it is the digital representation of a 11025 Hz sine wave whose amplitude is 1.414 Volt (+3 dBFS). And finally bring the signal into a D/A converter with a range of -1 Volt to +1 Volt. The converter will clip the signal, as it is designed to handle signals in the -1 Volt to +1 Volt range, but tries to reproduce a sine wave with an amplitude of 1.414 Volt.
This is the basic concept as far as I can understand. Maybe the full story is not so simple, but the principle should be similar to what I expressed.
Here is why oversampling D/A converters, with more bits than the original signal, are required. And here is why oversampling is required in digital signal processing. Oversampling the previous signal (+0.707, +0.707, -0.707, -0.707) by a factor of 2 yields the following samples: 0, +0.707, +1, +0.707, 0, -0.707, -1, -0.707. Here even a “simple” normalizing program can’t do wrong. The signal is already full-range. It cannot be further amplified.
I think that there is little knowledge about digital technology in general. Industry, of course, does not want to let everybody know the drawbacks of digital technology (in every field, not only in audio). I think that working in the digital domain adds one (or more) level(s) of indirection to anything. Quality and response times are sacrificed to the cheapness of using digital technology.
What are the implications of 0 dBFS+ in Media Center?
• Media Center should Analyze Audio using oversampling. This way peak level of more than 100% can be detected. The “amplified” signal in the previous example (+1, +1, -1, -1) should give a peak level as big as 141.4% (or +3 dBFS).
• Peak meters, if they’ll ever be implemented, should use oversampling as well.