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U. Glaeser

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8 ksamples/s for telephony (the signal spectrum extends up to 4 kHz, and thus covers most of the<br />

frequencies contained in speech), 32 ksamples/s for medium quality digital audio (audible frequency band<br />

up to 16 kHz is covered), 44.1 ksamples/s for a CD standard (audio frequency band up to 22.05 kHz is<br />

represented), 48 and 96 ksamples/s for high quality digital audio (the represented frequencies range up to<br />

24 and 48 kHz, respectively).<br />

An important generalization of the classic sampling theory applies to band signals [8]. A continuoustime<br />

signal, whose spectrum is limited to some frequency band<br />

© 2002 by CRC Press LLC<br />

(27.30)<br />

can be sampled with a sampling rate of at least F s = 2∆f only (i.e., critically sampled), if both spectrum<br />

border frequencies f 1 and f 2 in Eq. (27.30) are consecutive multiples of the Nyquist frequency F s/2, i.e., if<br />

(27.31)<br />

where k is an integer. Such a signal is referred to as the integer-band signal. Audio signals are not by themselves<br />

integer-band signals but they can be split with an analysis filter bank to some subband signals, which all<br />

are integer-band signals, and thus, can be critically sampled. This is indeed the case in many digital audio<br />

coders, e.g., in the MUSICAM standard the input audio signal, initially sampled with 48 ksamples/s, is split<br />

into 32 subbands with bandwidths of 24,000/32 = 750 Hz each. Signals in each subband are sampled with<br />

48/32 = 1.5 ksamples/s sampling rate.<br />

Another signal discretization process is quantization, i.e., the procedure of converting a signal with<br />

continuously distributed values into a signal with discrete values. Unlike sampling, which, under some<br />

conditions, can be considered lossless, i.e., the original signal can—at least theoretically—be perfectly<br />

recovered after sampling, quantization is an inherently lossy operation [8,37].<br />

The error due to the quantization has a nature of noise and is referred to as the quantization noise.<br />

Although this noise is unavoidable and cannot be removed from the signal, it can be made inaudible by<br />

controlling its level and forcing it to lie under the threshold of audibility. Masking effects, discussed in<br />

section 27.4, can be very effectively exploited with this end in view.<br />

The quantization noise is usually analyzed under the following simplifying assumptions:<br />

• The quantization steps are uniform.<br />

• The number of quantization levels is high.<br />

∆f = f1– f2, f2 > f1 f1 = kFs/2 and f2 = ( k + 1)Fs/2<br />

The first assumption is not fulfilled in many quantization techniques for audio signals. This is because<br />

the perception of noise does not depend on its absolute power but on the signal-to-noise ratio (SNR).<br />

Thus, it is reasonable to quantize audio signals nonuniformly, with quantization steps proportional to the<br />

signal values. If the steps are not uniform, then the quantization error will be a function of the input<br />

signal, and consequently, it will not be an additive noise any more. Fortunately, in most procedures for<br />

the quantization of audio signals, quantization steps are at least range by range uniform and the first<br />

assumption can be considered as approximately valid. The second assumption is usually satisfactorily<br />

fulfilled. Due to this assumption the quantization noise has a uniform probability density distribution<br />

and is not correlated with the signal.<br />

Denote by Q the quantization step and by p(x) the probability distribution function of the quantization<br />

error. Then<br />

∫<br />

Q/2<br />

p( x)dx<br />

=<br />

1<br />

– Q/2<br />

(27.32)

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