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The outer loop shapes the quantization noise according to the masking threshold for each subband. The system starts with a default scale factor of 1.0 for each subband. If the quantization noise in a given band exceeds the allowable noise, determined by the masking threshold supplied by the perceptual model, the scale factor for the band is adjusted to reduce the quantization noise. Since smaller quantization noise requires a larger number of quantization steps (and thus, a higher bit rate), the rate adjustment loop has to be repeated every time new scale factors are computed. In other words, the rate loop is nested within the noise control loop. The outer noise control loop executes until the actual noise, computed as the difference between the original spectrum values and the quantized spectrum values, is below the masking threshold for every subband. Compression techniques closely related to ISO/MPEG-1 Layer 3 audio compression find application in a number of popular streaming media formats. MPEG-2 AAC. Of particular note is MPEG-2 advanced audio coding (AAC), a highly advanced perceptual code, used initially for digital radio applications. The AAC code improves on previous techniques to increase coding efficiency. For example, an AAC system operating at 96 kilobits per second produces the same subjective sound quality as ISO/MPEG-1 Layer 2 operating at 192 kilobits per second; a 2:1 reduction in bit rate. The driving force to develop AAC was the quest for an efficient coding method for surround signals, like the five-channel systems (left, right, center, left-surround, right-surround) used in cinemas today. There have been algorithms for these signals in MPEG for quite a while, but optimum efficiency was not reached, for technical and historical reasons. The set aim, therefore, was a considerable decrease in the bit rate necessary to represent surround sound program material. There are three main modes (Profiles) in the AAC standard:
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Main Used when processing power and especially memory are readily available. Low complexity (LC) Used when processing cycles and memory use are constrained. Scaleable sampling rate (SSR) Appropriate when a scalable decoder is required. A scalable decoder can be designed to support different levels of audio quality from a common bit stream; for example, having both high- and low-cost implementations to support higher and lower audio qualities, respectively.
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Appendix E
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Different profiles trade off encoding complexity for audio quality at a given bit rate. For example, at 128 kilobits per second, the Main profile AAC code has a more complex encoder structure than the LC AAC code at the same bit rate, but provides better audio quality as a result. Like all perceptual coding schemes, MPEG-2 AAC makes use of the signal masking properties of the human auditory system in order to reduce the amount of data. In so doing, the quantization noise is distributed to frequency bands in such a way that it is masked by the aggregate signal (in other words, it remains inaudible). Even though the basic structure of this coding method hardly differs from its predecessors, there are some new aspects worth noting. A block diagram of the AAC system general structure is given in Figure E.18. The blocks in the drawing are referred to as tools that the coding algorithm uses to compress the digital audio signal. While many of these tools exist in most audio perceptual coders, two are unique to AAC: the temporal noise shaper (TNS) and the filterbank tool. The TNS uses a backward adaptive prediction process to remove redundancy between the frequency channels that are created by the filterbank tool. A true novelty in the area of time/frequency coding schemes, the TNS shapes the distribution of quantization noise in the time domain by prediction in the frequency domain. Prediction is a technique commonly used in speech coding systems. AAC benefits from the fact that many types of audio signal are easy to predict. Voice signals, in particular, experience considerable improvement through TNS.
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