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The Australian Broadcasting Corporation first used Dolby AC-l in 1985 for DBS applications. Partly because of its low decoder cost, AC-l has since been adopted for other DBS services, satellite communication networks, and digital cable radio systems; the data rate is 220 325 kilobits per second per channel depending on application. A refined form of adaptive delta modulation (ADM), the data stream contains information not on the absolute value of the audio signal, but on the change in value from sample to sample. Techniques adapted from Dolby noise reduction, such as continually varying step-size and pre-emphasis, improve on basic ADM performance. Dolby AC-2 uses advanced adaptive transform coding for professional audio applications; its data rate being 128 or 192 kilobits per second per audio channel. Frequency-domain signal processing in a multiplicity of narrow bands took full advantage of noise masking, resulting in data rate reduction combined with high signal transparency. Dolby noise reduction, an important precursor to the development of AC-3, works by lowering the noise when no audio signal is present, while allowing strong audio signals to cover or mask the noise at other times. Thus, it takes advantage of the psychoacoustic phenomenon known as auditory masking. Even when audio signals are present in some parts of the spectrum, Dolby NR reduces the noise in the other parts so the noise remains imperceptible. This is because audio signals can only mask noise that occurs at nearby frequencies. AC-3 was designed to take advantage of human auditory masking, in common with all other perceptual audio coding schemes. 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 is forced to stay very close in frequency to the frequency components of the audio signal being coded. Reducing or eliminating coding noise, wherever there are no audio signals to mask it, can subjectively preserve the sound quality of the original signal. In this respect, a perceptual coding system like AC-3 is essentially a form of very selective noise reduction. In AC-3, bits are distributed among the filter bands as needed by the particular frequency spectrum or dynamic nature of the program, as with AAC. 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
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Appendix E
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more data than sparsely occupied channels, for example, or strong sounds in one channel to provide masking for noise in other channels. AC-3 s masking model and its shared bit pool are key factors in its spectrum efficiency. Furthermore, where other coding systems use considerable (and precious) data to carry instructions to their decoders, AC3 can use proportionally more of the transmitted data to represent audio, which arguably means higher sound quality. AC-3 can process at least 20-bit dynamic range digital audio signals over a frequency range from 20 Hz to 20 kHz. The bass effects channel covers 20 to 120 Hz. Sampling rates of 32, 44.1 and 48 kHz are supported. Data rates in AC-3 coding range from as low as 32 kilobits per second, for a single mono channel, up to 640 kilobits per second, covering a wide range of applications. Typical applications include 384 kilobits per second for 5.1-channel Dolby Surround Digital consumer formats, and 192 kilobits per second for two-channel audio distribution. With AAC and AC-3 being similar in what they can do for the end-user, it is hard to see why one system should be preferred over the other. According to supporters of the MPEG-2 AAC standard, being a more recently developed, elaborate, and complex compression algorithm than the older AC-3, it provides superior compression efficiency and subjectively better sound quality. AAC and AC-3 are both transform coders, but AAC uses a filterbank with a finer frequency resolution that enables superior signal compression. AAC also uses a number of new tools, such as temporal noise shaping, backward adaptive linear prediction, joint stereo coding techniques, and Huffman coding of quantized components, each of which provides additional audio compression capability. Furthermore, AAC is much more flexible than AC-3, in that AAC supports a wide range of sampling rates and bit rates, from one to 48 audio channels, a multitude of low frequency enhancement channels, multilanguage capability, and support for embedded data streams. However, AC-3 has been cemented into the ATSC DTV standard and finds application in the highly successful domestic DVD standard, among other applications, so it isn t likely to vanish anytime soon.
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