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U. Glaeser

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where<br />

From Eqs. (27.32) and (27.33), the average quantization noise power p q can be calculated as<br />

The signal-to-noise ratio in decibels is then<br />

© 2002 by CRC Press LLC<br />

(27.33)<br />

(27.34)<br />

(27.35)<br />

where P s is the time-averaged signal power. Assume that the ADC has a full scale of m bits. Then the<br />

maximum input signal amplitude is<br />

and thus<br />

p( x)<br />

⎧1/Q<br />

for x∈[ – Q/2,Q/2 ]<br />

= ⎨<br />

⎩ 0 otherwise<br />

From Eqs. (27.34), (27.35), and (27.37) it follows that<br />

P q<br />

∫<br />

∞<br />

x 2 = p( x)dx<br />

= --- dx =<br />

Q<br />

– ∞<br />

(27.36)<br />

(27.37)<br />

(27.38)<br />

Thus, each additional bit in the quantized signal resolution means ca. 6 dB improvement in the SNR<br />

(or equivalently in the dynamic range). The “const” in expression (27.38) is of secondary importance.<br />

Its value depends on the signal probability density distribution and the ADC range. For instance, for an<br />

ADC range equal to ( – 4 Ps,4 Ps) , the respective value is const ≈ −7.3 dB.<br />

Representing a signal just as a stream of uniformly quantized samples is referred to as the pulse code<br />

modulation (PCM). Typical resolutions in bits per sample (bps) are 16 bps, 24 bps, and even 30 bps.<br />

For instance, for a CD standard with two stereo channels, 44.1 ksamples/s sampling rate and 16 bit<br />

resolution, the resulting audio bit rate is 2 × 44, 100 × 16 = 1.41 Mb/s. In reality, the CD standard has<br />

a large overhead bit rate due to 49-bit representation of every 16-bit sample. The resulting total bit rate<br />

is thus equal to (49/16) × 1.41 = 4.32 Mb/s.<br />

PCM representation is not an efficient method for high quality audio. In order to reduce the required<br />

bit rate, various data compression and coding techniques can be used. Simple but not very efficient<br />

approaches preserve the signal waveform and are therefore referred to as lossless coding techniques<br />

(section 27.8). Data compression facility of lossless audio coders is rather moderate. Average achievable<br />

bit per sample values are only slightly greater than 4.5 bps [35]. Sophisticated techniques, which are still<br />

subject of an intensive research, allow for a drastic reduction of this value—at least by one order of<br />

magnitude. These coding techniques are lossy in the sense that they corrupt the signal; however, this<br />

corruption, can be controlled in such a way that it is inaudible. Such audio coders are called transparent<br />

(section 27.9). In order to efficiently and transparently compress audio and/or speech, the knowledge<br />

about the speech and audio production (the parametric audio coding discussed in section 27.3) as well<br />

as the knowledge concerning the human auditory perception (discussed in section 27.4, resulting in the<br />

perceptual audio coding) should be exploited (Fig. 27.17).<br />

∫<br />

SNR = 10 log<br />

Q/2<br />

– Q/2<br />

x 2<br />

Ps ---- 10 Pq ⎛ ⎞<br />

⎝ ⎠<br />

A 2 m<br />

= ( – 1)Q<br />

P s<br />

2 m<br />

[ ( – 1)Q]<br />

2<br />

∝<br />

Q 2<br />

-----<br />

12<br />

SNR( m)<br />

= 20m log 2 + const ≈ 6.02m + const<br />

10

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