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Table 2-1 Speech Digitization Methods and Some Illustrative Examples
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Method Waveform coders Aspect
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Utilize algorithms to produce an output that approximates the input waveform Standard telephony method for toll-quality voice Typically used at 64 Kbps Adaptive coding for rates of 40, 32, 24, and 16 Kbps Uses a combination of adaptive quantization and adaptive prediction Digitizes a compact description of the voice spectrum in several frequency bands, including extraction of the pitch component of the signal Supports rates of 16 and 8 Kbps Speech is separated into frequency bands; each band is coded using different strategies The strategies are selected to suit properties of hearing and some predictive measure of the input spectrum Supports rates of 8 and 4 Kbps A suitable number of pulses are utilized to optimize the excitation information for a speech segment and supplement linear prediction of the segments Supports rates of 8 to 2 Kbps The coder stores a repository of candidate excitations, each a stochastic sequence of pulses, and the best is matched
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need 2W samples per second For voice, when band is limited to a nominal 4,000-Hz bandwidth, you need 8,000 samples per second The dynamic range of the signal (and, ultimately, the signal-to-noise ratio [SNR]) dictates the number of quantizing levels required For telephonic voice, 256 levels suffice based on psychoacoustic studies conducted in the 1950s and early 1960s It follows that 8 bits are needed to uniquely represent these many levels In turn, this implies that you need 64,000 bps to encode telephonic human speech in digital form PCM does not require sophisticated signalprocessing techniques and related circuitry; hence, it was the first method to be employed and is the prevalent method used today in the telephone plant (PCM was first deployed in the early 1960s) PCM provides excellent quality This is the method used in modern compact disc (CD) music recording technology, although the sampling rate is higher and the coding words are longer to guarantee a frequency response to 22 kHz The problem with PCM is that it requires a fairly high bandwidth to represent a voice signal Sophisticated voice-coding methods have become available in the past decade due to the evolution of very-large-scale integration (VLSI) technology Coding rates of 32 Kbps, 16 Kbps, and even vocoder methods requiring
Technologies for Packet-Based Voice Applications
6,300 bps, 5,300 bps, 4,800 bps, 2,400 bps, and even less have evolved in recent years while the quality of the synthesized voice has increased considerably There is interest for pursuing these new coding schemes, particularly, VoP, since the implication is that you can increase the voice-carrying capacity of the network in place up to 10 times without the introduction of new transmission equipment Unfortunately, current switching technology is based on DS0 (64-Kbps) channels As a rough figure, we estimate that there is an embedded base of about $50 billion of traditional Class 5 switches in North America Given this predicament, a carrier can either (a) ignore these new coding methods, (b) use trunk gateway hardware outside the switch to achieve the trunk-level voice compression, or (c) introduce new switching technology that uses these schemes directly Unfortunately, the poor quality of wireless telephony has made the traditional telephone network look great by comparison, so there is no uproar of demand from anyone to replace a working technology in the PSTN with another Figure 2-2 illustrates the target price per channel for an external gateway system that
Figure 2-2 Per-channel cost to achieve a 100 percent saving in transmission
Simple transmission (cost/mo) 8,000
4,000/mo 100% reduction 4,000 2,000/mo
1 Gbps 15,000
2 Gbps 30,000 2000
#DSO per trunk bundle 120,000 both ends,
New encoding equipment authorized over 60 months: $60 or $60,000 one end Cost per channel: $60,000/15,000 = $4
2
transcodes the trunk-side PCM voice between two switches to a new vocoder format and that aims at saving 100 percent of the transmission cost To achieve this goal, the cost per channel needs to be $4 In this example, the trunk cost is reduced by 100 percent (from $4,000 per month for a 15,000 DS0 trunk to $2,000) However, the entire saving is then applied to purchasing the new equipment; therefore, there is no net saving to the bottom line If you assume that the cost is reduced by 100 percent, and half of that saving is applied to the new equipment and the other half is accepted as bottom-line saving, the cost per channel on the gateway needs to be $2 Alternatively, as noted, the carrier could revamp the entire network, but the economic justification would then need to be strongly anchored on new revenues and applications One way to reduce the bit rate in a waveform-coding environment is to use differential encoding methods The problem with these voice-coding methods, however, is that if the input analog signal varies rapidly between samples, the differential technique is not able to represent the incoming signal with sufficient accuracy Just as in the PCM technique, clipping can occur when the input to the quantizer is too large; in this case, the input signal is the change in signal from the previous sample The resulting distortion is known as slope-overload distortion This issue is addressed by the adaptive differential pulse code modulation (ADPCM) scheme ADPCM provides toll-quality voice with minimal (voice) degradation at 32 Kbps In ADPCM, the coder can be made to adapt to slope overload by increasing the range represented by the 4 bits used per sample In principle, the range implicit in the 4 bits can be increased or decreased to match different situations; this will reduce the quantizing noise for large signals, but will increase noise for normal signals In practice, the ADPCM coding device accepts the PCM-coded signal and then applies a special algorithm to reduce the 8-bit samples to 4-bit words using only 15 quantizing levels These 4-bit words no longer represent sample amplitudes; instead, they contain only enough information to reconstruct the amplitude at the distant end The adaptive predictor predicts the value of the next signal based on the level of the previously sampled signal A feedback loop ensures that voice variations are followed with minimal deviation The deviation of the predicted value measured against the actual signal tends to be small and can be encoded with 4 bits In the event that successive samples vary widely, the algorithm adapts by increasing the range represented by the 4 bits through a slight increase in the noise level over normal signals12 LBRV methods such as ADPCM reduce not only the capacity needed to transmit digital voice, but also the capacity for voiceband data (such as fax and dialup Internet access) ADPCM encoding methods can and have been utilized in ATM and Frame Relay environments Vocoding is now used more prevalently in IP-based voice (Naturally, vocoding can also be used in Frame Relay environments)14
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