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-- DOLBY STEREO

Stereo

Let us consider a REAL stereo recording system (I mean not a simulated stereo based on mono sounds mixed together). It requires two microphones and provides two channels output. Two kinds of signals are recorded. Firstly direct sounds are recorded simultaneously on both channels as mono or in-phase signals. Nevertheless some direct sounds may reach only one microphone so that a signal occurs on one channel only (R, L or some blend). Secondly reverberations peculiar to the room or sounds coming from the rear of the virtual recording point will create indirect sounds. They reach microphones at different times and appear as out-of-phase signals between left and right channel. Thus stereo is obviously a natural two channel mixture of in-phase and out-of-phase signals, phase varying between 0 and 180 degrees. Reflections In True Stereo Dolby Stereo MP Matrix Encoding process Stereo gives depth and relief to the sound but it is absolutly not efficient for a movie. Indeed you cannot manage to record in stereo both dialogs (direct sounds) and ambiant sounds such as birds, sea or traffic jam without recording noise coming from the movie team for example. Moreover it would be impossible to locate sounds as the director wants to. Thus they use high directivity microphones to record the dialogs in mono. Then dialogs are mixed with music and different artificial sounds recorded in mono. This is done using the Dolby Matrix Encoding. Dolby laboratories use exactly the same process as natural Stereo except the fact that phase shift is either 0 or 180 degrees (anti-phase). If you listen to an un-decoded surround program on an ordinary stereo setup, you may detect the out-of-phaseness of surround signals. The soundstage may appear to be wider than your speakers, or you may actually have a psycho-acoustic experience of sound from behind you. Left and right channels go through the system without being altered. Center channel is equally added to both main channels to create an in-phase signal or mono signal. Finally surround channel is added with +90 degrees phase shift to left channel and -90 degrees to right channel. Optimally, a frequency dependent shift is probably used to create this phase shift. A pair of simple passive filter circuits might suffice. All signals are controled in gain by amplifiers circuits .Surround channel also employs a bandpass filter (100Hz/7KHz/-3dB) and a modified Dolby B noise reduction filter (5dB). Highly Simplified Dolby Stereo Encoder

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-- DOLBY DIGITAL

Left front channel Right front channel Left surround channel Right surround channel Center channel Subwoofer channel (.1)

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AC-3 has been designed to take maximum advantage of human auditory masking. It divides the audio spectrum of each channel into narrow frequency bands of different sizes optimized with respect to the frequency selectivity of human hearing. This makes it possible to sharply filter coding noise so that it isforced to stay very close in frequency to the frequency components of the audio signal being coded. By reducing or eliminating coding noise wherever there are no audio signals to mask it, the sound quality of the original signal can be subjectively preserved. In this key respect, a perceptual coding system like AC-3 is essentially a form of very selective and powerful noise reduction.

In Dolby AC-3, bits are distributed among the filter bands as needed by the particular frequency spectrum or dynamic nature of the program. A built-in model of auditory masking allows the coder to alter its frequency selectivity (as well as time resolution) to make sure that a sufficient number of bits are used to describe the audio signal in each band, thus ensuring noise is fully masked. AC-3 also decides how the bits are distributed among the various channels from a common bit pool. This technique allows channels with greater frequency content to demand more data than sparsely occupied channels, for example, or strong sounds in one channel to provide masking for noise in other channels.

Full details of the Dolby 5.1 surround system haven't been revealed, but how it generally works is as follows : The 6 source channels are digitized, at 20 bits/sample, and apparently at 44.1 KHz. The bandwidth allowed for each channel is 20Hz-20KHz. The subwoofer channel is low-pass filtered to eliminate sound above 300 Hz. The raw combined signal stream would be about 6 Mbps. Data compression is applied to eliminate redundancy, and psycho-acoustic masking rules are used to remove sounds which theoretically can't be heard anyway.The net result is a data stream of 384 Kbps/sec (about 1/4 of the standard audio CD rate) 16 bits encoded. The final soundtrack occupies the space between the sprocket holes at 320 Kbps. The picture below is a photograph of such a soundtrack.