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the analog signal). For linear PCM, the bit allocation is commonly 16. The level of each audio sample, therefore, will be compared with one of 216 or 65,536 discrete levels or steps. Compression or bit rate reduction of the PCM signal leads to the requantizing of an already quantized signal, which will unavoidably inject further quantization noise. It always has been good operating practice to restrict the number of A/D and D/A conversions in an audio chain. Nothing has changed in this regard and now the number of compression stages also should be kept to a minimum. Additionally, the bit rates of these stages should be set as high as practical; put another way, the compression ratio should be as low as possible. Sooner or later, after a finite number of A/D, D/A conversions and passes of compression coding, of whatever type the accumulation of quantization noise and other unpredictable signal degradations eventually will break through the noise/signal threshold, be interpreted as part of the audio signal, be processed as such, and be heard by the listener. Sampling Frequency and Bit Rate. is defined by:



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The rules regarding the selection of a sampling frequency are based on Nyquist s theorem [16]. This ensures that, in particular, the lower sideband of the sampling frequency does not fold into the base-band audio. Objectionable and audible aliasing effects would occur if the two bands were to overlap. In practice, the sampling rate is set slightly above twice the highest audible frequency, which makes filter designs less complex and less expensive. (In fact, the setting of the sampling frequency at slightly more than twice the audio base-bandwidth is a historical artifact, since most conversion systems today employ oversampling, which enables very precise definition of the audio pass-band through digital anti-alias filtering at very low cost. When anti-aliasing filters were analog, they were both imprecise and expensive.) In the case of a stereo CD with the audio signal having been sampled at 44.1 kHz, this sampling rate produces audio bandwidths of approximately 20 kHz for each channel. The resulting audio bit rate 44.1 kHz 16 2 1.411 megabits per second, as discussed previously.





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You should also use the xml:lang attribute inside the xsd:documentation element to indicate the language in which your comment is written. Annotations can be placed only: Before and after any global component (such as schema, simpleType, and attribute) At the beginning of nonglobal components

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Most audio compression systems are based upon one of two basic technologies [16]:

Predictive or adaptive differential PCM (ADPCM) time-domain coding Transform or adaptive PCM (APCM) frequency-domain coding

It is in their approaches to dealing with the redundancy and irrelevancy of the PCM signal that these techniques differ. The time domain or prediction approach includes G.722, which has been a universal standard since the mid-70s and was joined by a proprietary algorithm, apt-X100. Both these algorithms deal mainly with redundancy. The frequency domain or transform method adopted by a number of algorithms deal in irrelevancy, adopting psychoacoustic masking techniques to identify and remove those unwanted sounds. This range of algorithms include the industry standards ISO.MPEG-1 Layers 1, 2, and 3; apt-Q; MUSICAM; Dolby AC-2 and AC-3, and others. Subband Coding. Without exception, all of the algorithms mentioned in the previous section process the PCM signal by splitting it into a number of frequency subbands, in one case as few as two (G.722) or as many as 1024 (apt-Q) [15]. MPEG-1 Layer 1, with 4:1 compression, has 32 frequency subbands and is the system found in the now rare Digital Compact Cassette (DCC). The MiniDisc ATRAC proprietary algorithm at 5:1 has a more flexible multi-subband approach, which is dependent on the complexity of the audio signal. Subband coding enables the frequency domain redundancies within the audio signals to be exploited. This permits a reduction in the coded bit rate, compared to PCM, for a given signal fidelity. Spectral redundancies are also present as a result of the signal energies in the various frequency bands being unequal at any instant of time. By altering the bit allocation for each subband, either by dynamically adapting it according to the energy of the contained signal or by fixing it for each subband, the quantization noise can be reduced across all bands. This process compares favorably with the noise characteristics of a PCM coder performing at the same overall bit rate.

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