In this treatise a range of Line Spectrum Frequency (LSF) Vector Quantization (VQ) schemes were studied comparatively, which were designed for wideband speech codecs. Both predictive arrangements and memoryless schemes were investigated. Specifically, both memoryless Split Vector Quantization (SVQ) and Classified Vector Quantization (CVQ) were studied. These techniques exhibit a low complexity and high channel error resilience, but require high bit rates for maintaining high speech quality. By contrast, Predictive Vector Quantizers (PVQ) offer an enhanced Spectral Distortion (SD) performance, although they are sensitive to channel error propagation. It is shown that the family of so—called Safety—Net Vector Quantization (SNVQ) schemes offers a good design compromise, providing an extension to memory—based PVQ, and thereby improving the performances both over noisy and noiseless channels.

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A burst-by-burst adaptive speech transceiver is proposed, which can drop its source coding rate and speech quality under transceiver control in order to invoke a more error resilient modem mode amongst less favourable channel conditions. The novel, high-quality, Adaptive Multi-Rate (AMR) speech codec [5], operated at bit rates of 4.75 and 10.2 kbps and combined with sourcesensitivity-matched Redundant Residue Number Systems (RRNS) based channel codes. Burst-by-burst adaptive Joint-Detection based Code-Division Multiple Access (JD-CDMA) is used for transmitting the dual-rate bitstream generated by the AMR speech codec. SYSTEM OVERVIEWIn recent years the concept of near-instantaneously adaptive transceivers has reached a state of maturity and various adaptive features have found their way in to standard systems [1]. The time-variant quality fluctuations of the mobile channel result in a time-variant bitrate [2] and hence special attention has to be devoted to contriving interactive multimedia systems, which are capable of efficiently accommodating these fluctuating bit rates [3,4]. In this contribution, we propose a dual-mode burst-by-burst adaptive speech transceiver scheme, based on the Advanced Multi Rate (AMR) speech codec [5,6], Redundant Residue Number System (RRNS) assisted channel coding [7] and Joint Detection aided Code-Division Multiple Access (JD-CDMA) [8]. The schematic of the proposed adaptive JD-CDMA speech transceiver is depicted in Figure 1. The mode switching is controlled by the channel quality fluctuations imposed by the time-variant channel. This is not a desirable scenario. However, we will endeavour to contrive measures in order This work has been performed in the framework of the Pan-Euroepan IST project IST-1999-12070 (TRUST), which is partly funded by the European Union. The authors would like to acknowledge the contributions of their colleagues, although the views expressed are those of the authors.The financial support of the EPSRC, Swindon, UK is also gratefully acknowledged.Globecom '2000, San Francisco, USA, 27 Nov. -1 Dec. 2000 to mitigate the associated perceptual speech quality fluctuations. The underlying trade-offs associated with employing two speech modes of the AMR standard speech codec in conjunction with a reconfigurable, unequal error protection BPSK/4QAM modem are investigated. THE AMR SPEECH CODECThe AMR codec employs the Algebraic Code-Excited Linear Predictive (ACELP) model [9]. Here we provide a brief overview of the AMR codec following the approach of [5,6]. The AMR codec's complexity is relatively low and hence it can be implemented cost-efficiently. This codec operates on a 20ms frame of 160 speech samples, and generates encoded blocks of 95, 103, 118, 134, 148, 159, 204 and 244 bits/20ms. This leads to bit rates of 4.75, 5.15, 5.9, 6.7, 7.4, 7.95, 10.2 and 12.2 kbps, respectively. Explicitly, the AMR speech codec provides eight different modes. Multirate coding [10] allows a variation in the total allocation of bits for a speech frame, adapting the rate t...

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