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This asymmetrical limiting has the effect of delaying the start of a fastto slow-state transition until the absolute value of Ic(k) remains constant for some time This tends to eliminate premature transitions for pulsed input signals such as switched carrier voiceband data Adaptive Predictor and Feedback Reconstructed Signal Calculator The primary function of the adaptive predictor is to compute the signal estimate se(k) from the quantized difference signal dq(k) Two adaptive predictor structures are used: a sixth-order section that models zeros and a second-order section that models poles in the input signal This dual structure effectively caters for the variety of input signals that might be encountered The signal estimate is computed by the following: se 1k2 a1 1k 12sr 1k 12 a2 1k 12sr 1k 22 sez 1k2
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and the reconstructed signal is defined as sr 1k i2 se 1k i2 dq 1k i2
Both sets of predictor coefficients are updated using a simplified gradient algorithm Tone and Transition Detector In order to improve performance for signals originating from FSK modems operating in the character mode, a twostep detection process is defined First, partial-band signal (that is, tone) detection is invoked so that the quantizer can be driven into the fast mode of adaptation: td 1k2
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0, otherwise In addition, a transition from a partial band is defined so that the predictor coefficients can be set to zero and the quantizer can be forced into the fast mode of adaptation:
tr 1k2
1, if a2 1k2 6
071875 and 0 dq 1k2 0 7 24
2yl1k2
0, otherwise
233 ADPCM Decoder Principles
Figure 2-13 is a block schematic of the decoder There is a feedback path and a feedforward path The feedback path uses the core bits to calculate the signal estimate The feedforward path contains the core and enhanced bits, and reconstructs the output PCM codeword
Technologies for Packet-Based Voice Applications
Output PCM sp(k) Synchronous sd(k) format coding conversion adjustment
Figure 2-13 Block schematic of the decoder
Feedback BIT masking
s(k)
sr(k)FF Feedforward dq(k)FF Feedforward inverse adaptive reconstructed quantizer signal calculator y(k)
I(k)
ADPCM input
Ic(k)
dq(k)FB Feedback Feedback sr(k)FB inverse adaptive reconstructed quantizer signal calculator se(k)
Adaptive predictor
B2(k)
Quantizer scale factor adaptation
y(k) yl(k) a1(k)
Adaptation speed control
tr(k)
td(k)
Tone and transition detector
234 Example of Application
For intercontinental connections, the use of ADPCM at 32 or 40 Kbps for improved voice transmission efficiency has become commonplace ITU-T standards for ADPCM support about the same bandwidth as PCM, but provide a reduced SNR: about 21 dB at 32 Kbps (G721) or about 28 dB at 40 Kbps (G726) Proprietary 32-Kbps ADPCM encoders/decoders (codecs) that support a reduced bandwidth of less than 3,200 Hz at an SNR of about 28 dB are also in common use96 Use of this technology for VoP networks is also possible, although not all that common
24 Technology and Standards for LBRV Methods
As noted in the previous sections, during the past quarter century there has been a significant level of research and development in the area of vocoder technology and compressed speech During the early to middle 1990s, the ITU-T (specifically SG 14 and SG 15) standardized several vocoders that are applicable to low-bit-rate multimedia communications in general, and VoP in intranets, Internet, and private-label IP networks in particular
2
Standardization is critical for interoperability and the assurance of ubiquitous end-to-end connectivity The recent standards are G728, G729, G729A, and G7231, as listed in Table 2-2 For some applications, the dominant factor is cost; for other applications, quality is paramount This is part of the reason why several standards have evolved in the recent past However, to be ultimately successful, VoP will have to narrow down to one choice (or a small set of choices) so that anyone can call anyone else (as we do today with modems or telephone instruments), without worrying what technology the destination party may be using Corporate enterprise networks and intranets are chronically congested Hence, for VoIP to take off, you must trade off high desktop computational power for compressing speech down to the lowest possible rates, to keep congestion low, with the delay budget Excessive delay introduces major quality-impacting artifacts The vocoders discussed in the rest of this chapter require between 10 and 20 million of instructions per second (MIPS) When contemplating running these on a desktop PC or a chipset, it is worth noting that a 226-MHz Pentium II runs at 560 MIPS See Table 2-6 and Figure 2-14 for a sample of the computing power of processors (Vocoders are typically implemented in digital signal processing chips, but Table 2-6 and Figure 2-14 provide an intuitive sense of the required computing power) This discussion focuses on G729, G729A, and G7231; G728 is also covered, but its data rate (16 Kbps) may be too high for (enterprise) VoP applications (although it would not necessarily be too high for carrier applications) ITU-T Recommendation G729 is an 8-Kbps conjugate-structure algebraic code-excited linear prediction (CS-ACELP) speech algorithm providing good speech quality G729 was originally designed for wireless environments, but it is applicable to IP/multimedia communications as well Annex A of ITU-T Recommendation G729 (also called G729A) describes a reduced-complexity version of the algorithm that has been designed explicitly for integrated voice and data applications that are prevalent in SoHo
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