Pengkodean Suara

15.1 Pendahuluan 15.1.1 Komunikasi telefon sebagai layanan multimedia percakapan When O.Nubaumer telah sukses menjadi org pertama yang mentransmisikan percakapan dan musik secara wireless dalam percobaan di Graz of Technology pada 1904, tidak ada seorang pun yang dapat memprediksi perkembangan hebat apa yang akan terjadi pada 100 tahun yang akan datang setelah prestasi dalam dunia komunikasi multimedia wireless ini tercapai. Beberapa tipe media baru sudah muncul seperti text, gambar, video, dan layanan modern dari komunikasi satu arah sampai komunikasi dua arah. Namun tetap, komunikasi telepon menjadi tulang punggung semua layanan percakapan dan terus berlanjut menjadi fungsi yang sangat dibutuhkan oleh terminal komunikasi bergerak. Cerita sukses mengenai pengkodean percakapan secara digital dimulai dengan perkenalan teknologi digital switching pada PSTN dengan menggunakan modulasi PCM 64 kbps dan berlanjut dengan standar kompresi yang canggih dari 32kbps di awal tahun 1980, diatas 16kbps sampai 8kbps di akhir tahun 1990. Untuk telefon wireless, persyaratan pada pengkodean digital cukup ketat dengan memperhatikan kecepatan data 15.1.2 Dasar Pengkodean Dasar pengkodean yang telah ditemukan oleh C. Shannon, orang yang mengembangkan tidak hanya teori pengkodean kanal tetapi juga teori menghitung distorsi untuk kompresi sinyal. The foundations for source coding were laid by C. Shannon [1959], who developed not only channel-coding theory for imperfect transmission channels but alsoratedistortion theoryfor signal compression. The latter theory is based on two components: a stochastic source model which allows us to characterize theredundancy in source information; and a distortion measure which characterizes therelevance of source information for a user. For asymptotically infinite delay and complexity, and certain simple source models and distortion measures, it can be shown that there exists an achievable lower bound on the bit rate necessary to achieve a given distortion level and, vice versa, that their exists an achievable lower bound on the distortion to be tolerated for a given bit rate. While complexity is an ever-dwindling obstacle, delay is a substantial issue in telephony, as it degrades the interactive quality of conversations severely when it exceeds a few 100 ms. Therefore, the main insight from rate distortion theory is the existence of a three-way tradeoff among the fundamental parameters rate, distortion,and delay. Traditional telephony networks operate in circuit-switched mode where transmission delay is essentially given by the electromagnetic propagation time and becomes only noticeable when dealing with satellite links. However, packet-switched networks are increasingly being used for telephony as well as in Voice over Internet Protocol (VoIP) systems where substantial delays can be accumulated in router queues, etc. In such systems, delay becomes the most essential parameter and will determine the achievable rate distortion tradeoff. Source coding with a small but tolerable level of distortion is also known aslossy coding whereas the limiting case of zero distortion is known aslossless coding. In most cases, a finite rate allows lossless coding only for discrete amplitude signals which we might consider fortranscoding of PCM speech i.e., the digital compression of speech signals which have already been digi-

tized with a conventional PCM codec. However, for circuit-switched wireless speech telephony, such lossless coders have two drawbacks: first, they waste the most precious resource i.e., the allocated radio spectrum as they invest more bits than necessary to meet the quality expectations of a typical user; second, they often result in a bitstream with a variable rate e.g., when using a Huffman coder which cannot be matched efficiently to thefixed rateoffered by circuitswitched transmission. Variable-rate coding is, however, a highly relevant topic in packet-switched networks and in certain applications of joint source channel coding for circuit-switched networks (see Section 15.4.5). While Shannons theory shows that, under idealized conditions, source coding and channel coding can be fully separated such that the two coding steps can be designed and optimized independently, this is not true under practical constraints such as finite delay or time-varying conditions where only a joint design of the source and channel coders is optimal. In this case, a fixed rate offered by the network can be advantageously split into a variable source rate and a variable channel code rate.