Jason Woodard

Comparative study of turbo decoding techniques: an overview

We provide an overview of the novel class of channel codes referred to as turbo codes, which have been shown to be capable of performing close to the Shannon limit. We commence with a discussion on turbo encoding, and then move on to describing the form of the iterative decoder most commonly used to decode turbo codes. We then elaborate on various decoding algorithms that can be used in an iterative decoder, and give an example of the operation of such a decoder using the so-called soft output Viterbi (1996) algorithm (SOVA). Lastly, the effect of a range of system parameters is investigated in a systematic fashion, in order to gauge their performance ramifications.

Voice and Audio Compression for Wireless Communications

About the Authors. Other Wiley and IEEE Press Books on Related Topics. Preface and Motivation. Acknowledgements. I Speech Signals andWaveform Coding. 2 Predictive Coding. 3 Analysis-by-synthesis Principles. 4 Speech Spectral Quantization. 5 RPE Coding. 6 Forward-Adaptive CELP Coding. 7 Standard CELP Codecs. 8 Backward-Adaptive CELP Coding. 9 Wideband Speech Coding. 10 MPEG-4 Audio Compression and Transmission. 11 Overview of Low-rate Speech Coding. 12 Linear Predictive Vocoder. 13 Wavelets and Pitch Detection. 14 Zinc Function Excitation. 15 Mixed-Multiband Excitation. 16 Sinusoidal Transform Coding Below 4kbps. 17 Conclusions on Low Rate Coding. 18 Comparison of Speech Transceivers. 19 Voice Over the Internet Protocol. A Constructing the Quadratic Spline Wavelets. B Zinc Function Excitation. C Probability Density Function for Amplitudes. Bibliography. Index. Author Index.

Turbo Decoding and Detection for Wireless Applications

A historical perspective of turbo coding and turbo transceivers inspired by the generic turbo principles is provided, as it evolved from Shannons visionary predictions. More specifically, we commence by discussing the turbo principles, which have been shown to be capable of performing close to Shannons capacity limit. We continue by reviewing the classic maximum a posteriori probability decoder. These discussions are followed by studying the effect of a range of system parameters in a systematic fashion, in order to gauge their performance ramifications. In the second part of this treatise, we focus our attention on the family of iterative receivers designed for wireless communication systems, which were partly inspired by the invention of turbo codes. More specifically, the family of iteratively detected joint coding and modulation schemes, turbo equalization, concatenated space-time and channel coding arrangements, as well as multi-user detection and three-stage multimedia systems are highlighted.

An intelligent multimode voice communications system for indoor communications

A novel high-quality, low-complexity dual-rate 4.7 and 6.5 kbits/s algebraic code excited linear predictive codec is proposed for adaptive multi-mode speech communicators, which can drop their source rate and speech quality under network control in order to invoke a more error resilient modem amongst less favorable channel conditions. Source-matched binary Bose-Chaudhuri-Hocquenghem (BCH) codecs combined with unequal protection diversity- and pilot-assisted 16and 64-level quadrature amplitude modulation (16-QAM, 64-QAM) are employed in order to accommodate both the 4.7 and the 6.5 kbits/s coded speech bits at a signaling rate of 3.1 kBd. Assuming an excess bandwidth of 100%, in a bandwidth of 200 kHz 32 time slots can be created, which allows us to support in excess of 50 users, when employing packet reservation multiple access (PRMA). Good communications quality speech is delivered in an equivalent bandwidth of 4 kHz, if the channel signal-to-noise ratio (SNR) and signal-to-interference ratio (SIR) of the benign indoors cordless channel are in excess of about 15 and 25 dB for the lower and higher speech quality 16-QAM and 64-QAM systems, respectively, and the PRMA time-slots are sufficiently uninterfered due to using time-slot classification algorithms and due to the attenuation of partitioning walls and ceilings. >

Turbo-coded parallel modem techniques for personal communications

We study the performance of the wideband orthogonal frequency division multiplexing (OFDM) system in conjunction with channel coding based on turbo codes over a range of wideband Rayleigh fading channels. Due to their diversity effect wideband propagation channels provide similar gains for OFDM modems to those of equalised narrowband channels, resulting in substantial coding gains, when combined with turbo coding.

A low-delay multimode speech terminal

The intelligent, adaptively reconfigurable wireless systems of the near future require programmable source codecs in order to optimally configure the transceiver to adapt to time-variant channel and traffic conditions. Hence we developed a programmable 8-16 kbits/s low-delay speech codec, which is compatible with the G728 16 kbits/s ITU codec at its top rate and offers a graceful trade-off between the speech quality and bit rate in the 8-16 kbits/s range. The issues of robustness against channel errors strongly influenced the algorithmic design of the 8-16 kbits/s speech codec, and hence special attention is devoted to these issues. Source-matched Bose-Chaudhuri-Hocquenghem (BCH) codecs combined with unequal protection pilot-assisted 4- and 16-level quadrature amplitude modulation (4-QAM, 16-QAM) are employed in order to transmit both the 8 and the 16 kbits/s coded speech bits at a signalling rate of 10.4 kBd. In a bandwidth of 1728 kHz, which is used by the Digital European Cordless Telephone (DECT) system 55 duplex or 110 simplex time slots can be created. Good toll quality speech is delivered in an equivalent bandwidth of 15.71 kHz, if the channel signal-to-noise ratio (SNR) and signal-to-interference ratio (SIR) are in excess of about 18 and 26 dB for the lower and higher speech quality 4-QAM and 16-QAM modes, respectively.

Automatic repeat request assisted cordless telephony

A generic re-transmission scheme is proposed for packet reservation multiple access (PRMA) assisted indoors cordless voice communications. Its frame error rate (FER) versus channel signal-to-noise ratio (SNR) performance is assessed under various traffic loading conditions using the example of a re-configurable transceiver. A significant operating SNR reduction can be achieved, if the traffic load is not excessive. This effect is particularly pronounced, if the speech transceiver used can drop its transmission rate and accommodate this lower rate using a more robust modem, when attempting re-transmission. The key system features are summarised.

An intelligent dual-mode wireless speech transceiver

Publisher Summary This chapter discusses the dual-mode wireless speech transceiver. The chapter investigates the underlying trade-offs of using a dual-rate, algebraic-code excited linear-predictive (ACELP) speech codec in conjunction with a diversity- and pilot-assisted coherent, reconfigurable, unequal error protection 16-QAM/64-QAM modem. It briefly describes the reconfigurable transceiver scheme and provides details of dual-rate ACELP codec. It also provides a short discussion on bit-sensitivity issues and source-matched embedded error protection. Packet reservation multiple access (PRMA) is described. This chapter closes with the characterization of system performance. It highlights the code design approach using the 4.7 kbps codec and notes that similar principles were followed in case of the 6.5 kbps codec. The reconfigurable transceiver has a single-user rate of 3.1 kBd and can accommodate 32 PRMA slots at a PRMA rate of 100 kBd in a bandwidth of 200 kHz. The number of users supported is in excess of 50 and the minimum channel SNR for the lower speech-quality mode is about 15 dB, while it about 25 dB for the higher quality mode. The number of time slots can be further increased to 42, when opting for a modulation access bandwidth of 50%, accommodating a signaling rate of 133 kBd within the 200 kHz system bandwidth. This will inflict a slight bit-error rate penalty, but pay dividends in terms of increasing the number of PRMA users by about 20.

A Range of Low and High Delay CELP Speech Codecs Between 8 and 4 kbits/s

Abstract In this paper we study the performance and the error sensitivities of six CELP [1] based codecs operating between 8 and 4 kbits/s. Codecs using both forward and backward adaption of the linear prediction coefficients and the long term predictor (LTP) are described. Initially we describe four low delay codecs which all use backward adaption of the LPC coefficients but which differ in their use of LTP. These codecs all have frame-lengths of 3 ms or less, and their performance at various bit rates between 8 and 4 kbits/s is examined. Next the error sensitivity of these codecs, and means of improving it, are described. Then an algebraic CELP (ACELP) [2] codec operating at 6.2 kbits/s with a frame-length of 5 ms is described. Our final codec also uses ACELP and operates between 4.7 and 7.1 kbits/s, but it is forward adaptive and so it has a much longer frame-length of up to 30 ms. After describing this codec we compare the performance of our codecs in both error-free conditions and in the presence of channel errors. Surprisingly the error sensitivity of the low delay backward adaptive codec with no LTP is similar to that of the forward adaptive, high delay, ACELP codec.

Performance and error sensitivity comparison of low and high delay CELP codecs between 8 and 4 kbits/s

In this paper we study the performance and error sensitivities of several CELP based codecs operating between 8 and 4 kbits/s. Both forward and backward adaption techniques are used for the short and the long term predictors, and both trained and algebraic excitation codebooks are used. Three codecs which employ backward adaption of their short term predictors and operate with frame-lengths of 3 ms or less are described. These codecs operate between 8 and 4 kbits/s and their performance at these bit rates is compared to a traditional forward adaptive algebraic CELP codec operating at 4.7, 6.5 and 7.1 kbits/s. Furthermore, the error sensitivity of the backward adaptive codecs, and means of improving this error sensitivity, are investigated. Finally, we compare the error sensitivity of the low delay, backward adaptive codecs to the high delay, forward adaptive codecs. Surprisingly, we found that it is possible to achieve good error resilience, comparable to that of the forward adaptive codec, using low delay backward adaptive codecs.