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Voice Digitization via Pulse Code Modulation
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One of the first processes in the transmission of a telephone call is the conversion of an analog signal into a digital one. This process is called pulse
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Downloaded from Digital Engineering Library @ McGraw-Hill (www.digitalengineeringlibrary.com) Copyright 2004 The McGraw-Hill Companies. All rights reserved. Any use is subject to the Terms of Use as given at the website.
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code modulation (PCM). This is a four-step process consisting of pulse amplitude modulation (PAM) sampling, companding, quantization, and encoding. Pulse Amplitude Modulation (PAM) The first stage in PCM is known as PAM. In order for an analog signal to be represented as a digitally encoded bitstream, the analog signal must be sampled at a rate that is equal to twice the bandwidth of the channel over which the signal is to be transmitted. As each analog voice channel is allocated 4 kHz of bandwidth, each voice signal is sampled at twice that rate, or 8,000 samples per second. In a T-Carrier, the standard in North America and Japan, each channel is sampled every one eight-thousandth of a second in rotation, resulting in the generation of 8,000 pulse amplitude samples from each channel every second. If the sampling rate is too high, too much information is transmitted and bandwidth is wasted. If the sampling rate is too low, aliasing may result. Aliasing is the interpretation of the sample points as a false waveform due to the lack of samples. Companding The second process of PCM is companding. Companding is the process of compressing the values of the PAM samples to fit the nonlinear quantizing scale that results in bandwidth savings of more than 30 percent. It is called companding as the sample is compressed for transmission and expanded for reception.5 Quantization The third stage in PCM is quantization. In quantization, values are assigned to each sample within a constrained range. In using a limited number of bits to represent each sample, the signal is quantized. The difference between the actual level of the input analog signal and the digitized representation is known as quantization noise. Noise is a detraction to voice quality and it is necessary to minimize noise. The way to do this is to use more bits, thus providing better granularity. In this case, an inevitable trade-off takes place bewteen bandwidth and quality. More bandwidth usually improves signal quality, but bandwidth costs money. Service providers, whether using TDM or Voice over IP (VoIP) for voice transmission will always have to choose between quality and bandwidth. A process known as nonuniform quantization involves the usage of smaller
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Shepard, Steven. SONET/SDH Demystified. New York: McGraw-Hill, 2001. p. 15 21.
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quantization steps at smaller signal levels and larger quantization steps for larger signal levels. This gives the signal greater granularity or quality at low signal levels and less granularity (quality) at high signal levels. The result is to spread the signal-to-noise ratio more evenly across the range of different signals and to enable fewer bits to be used compared to uniform quantization. This process results in less bandwidth being consumed than for uniform quantization.6 Encoding The fourth and final process in PCM is encoding the signal. This is performed by a codec (coder/decoder). Three types of codecs exist: waveform codecs, source codecs (also known as vocoders), and hybrid codecs. Waveform codecs sample and code an incoming analog signal without regard to how the signal was generated. Quantized values of the samples are then transmitted to the destination where the original signal is reconstructed, at least to a certain approximation of the original. Waveform codecs are known for simplicity with high-quality output. The disadvantage of waveform codecs is that they consume considerably more bandwidth than the other codecs. When waveform codecs are used at low bandwidth, speech quality degrades markedly. Source codecs match an incoming signal to a mathematical model of how speech is produced. They use the linear predictive filter model of the vocal tract, with a voiced/unvoiced flag to represent the excitation that is applied to the filter. The filter represents the vocal tract and the voice/unvoiced flag represents whether a voiced or unvoiced input is received from the vocal chords. The information transmitted is a set of model parameters as opposed to the signal itself. The receiver, using the same modeling technique in reverse, reconstructs the values received into an analog signal. Source codecs also operate at low bit rates and reproduce a synthetically sounding voice. Using higher bit rates does not result in improved voice quality. Vocoders (source codecs) are most widely used in private and military applications. Hybrid codecs are deployed in an attempt to derive the benefits from both technologies. They perform some degree of waveform matching while mimicking the architecture of human speech. Hybrid codecs provide better voice quality at low bandwidth than waveform codecs. Table 2-1 provides an outline of the different ITU codec standards and Table 2-2 lists the parameters of the voice codecs.
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