This paper presents an evaluation of the QoS enhancements to the IEEE 802.11 standard, named IEEE 802.11e, currently under specification. Both the enhanced distributed coordination function (EDCF) and hybrid coordination function (HCF) modes of medium access control (MAC) operation are analysed and compared with legacy distributed coordination function (DCF) and point coordination function (PCF). Performance evaluation is attained through computer simulation of a scenario of 802.11 b/e access to an IP core network through an access point (AP) in an infrastructure WLAN.

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Performance evaluation of IEEE 802.11e

Single- and Multi-carrier Quadrature Amplitude Modulation Principles and Applications for Personal Communications, WLANs and Broadcasting L. Hanzo Department of Electronics and Computer Science, University of Southampton, UK W. Webb Motorola, Arlington Heights, USA formerly at Multiple Access Communications Ltd, Southampton, UK T. Keller Ubinetics, Cambridge Technology Centre, Melbourn, UK formerly at Department of Electronics and Computer Science, University of Southampton, UK Motivated by the rapid evolution of wireless communication systems, this expanded second edition provides an overview of most major single- and multi-carrier Quadrature Amplitude Modulation (QAM) techniques commencing with simple QAM schemes for the uninitiated through to complex, rapidly-evolving areas, such as arrangements for wide-band mobile channels. Targeted at the more advanced reader, the multi-carrier modulation based second half of the book presents a research-orientated outlook using a variety of novel QAM-based arrangements. * Features six new chapters dealing with the complexities of multi-carrier modulation which has found applications ranging from Wireless Local Area Networks (WLAN) to Digital Video Broadcasting (DVB) * Provides a rudimentary introduction for readers requiring a background in the field of modulation and radio wave propagation * Discusses classic QAM transmission issues relevant to Gaussian channels * Examines QAM-based transmissions over mobile radio channels * Incorporates QAM-related orthogonal techniques, considers the spectral efficiency of QAM in cellular frequency re-use structures and presents a QAM-based speech communications system design study * Introduces Orthogonal Frequency Division Multiplexing (OFDM) over both Gaussian and wideband fading channels By providing an all-encompassing self-contained treatment of single- and multi- carrier QAM based communications, a wide range of readers including senior undergraduate and postgraduate students, practising engineers and researchers alike will all find the coverage of this book attractive.

Single- and Multi-carrier Quadrature Amplitude Modulation : Principles and Applications for Personal Communications, WLANs and Broadcasting

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Quadrature Amplitude Modulation: From Basics to Adaptive Trellis-Coded, Turbo-Equalised and Space-Time Coded OFDM, CDMA and MC-CDMA Systems

A speech coding system employing an adaptive codebook model of periodicity is augmented with a pitch-predictive filter (PPF). This PPF has a delay equal to the integer component of the pitch-period and a gain which is adaptive based on a measure of periodicity of the speech signal. In accordance with an embodiment of the present invention, speech processing systems which include a first portion comprising an adaptive codebook and corresponding adaptive codebook amplifier and a second portion comprising a fixed codebook coupled to a pitch filter, are adapted to delay the adaptive codebook gain; determine the pitch filter gain based on the delayed adaptive codebook gain, and amplify samples of a signal in the pitch filter based on said determined pitch filter gain. The adaptive codebook gain is delayed for one subframe. The pitch filter gain equals the delayed. adaptive codebook gain, except when the adaptive codebook gain is either less than 0.2 or greater than 0.8., in which cases the pitch filter gain is set equal to 0.2 or 0.8, respectively.

CS-ACELP speech compression system with adaptive pitch prediction filter gain based on a measure of periodicity

An aspect of the present invention relates to a method for low-frequency emphasizing the spectrum of a sound signal transformed in a frequency domain and comprising transform coefficients grouped in a number of blocks, in which a maximum energy for one block is calculated and a position index of the block with maximum energy is determined, a factor is calculated for each block having a position index smaller than the position index of the block with maximum energy, and for each block a gain is determined from the factor and is applied to the transform coefficients of the block.

Methods and devices for low-frequency emphasis during audio compression based on acelp/tcx

This paper describes the adaptive multirate wideband (AMR-WB) speech codec selected by the Third Generation Partnership Project (3GPP) for GSM and the third generation mobile communication WCDMA system for providing wideband speech services. The AMR-WB speech codec algorithm was selected in December 2000 and the corresponding specifications were approved in March 2001. The AMR-WB codec was also selected by the International Telecommunication Union-Telecommunication Sector (ITU-T) in July 2001 in the standardization activity for wideband speech coding around 16 kb/s and was approved in January 2002 as Recommendation G.722.2. The adoption of AMR-WB by ITU-T is of significant importance since for the first time the same codec is adopted for wireless as well as wireline services. AMR-WB uses an extended audio bandwidth from 50 Hz to 7 kHz and gives superior speech quality and voice naturalness compared to existing second- and third-generation mobile communication systems. The wideband speech service provided by the AMR-WB codec will give mobile communication speech quality that also substantially exceeds (narrowband) wireline quality. The paper details AMR-WB standardization history, algorithmic description including novel techniques for efficient ACELP wideband speech coding and subjective quality performance of the codec.

The adaptive multirate wideband speech codec (AMR-WB)

The application of algebraic code excited linear prediction (ACELP) coding to wideband speech is presented An algebraic codebook with a 20 bit address can be used without any storage requirements and, more importantly, with a very efficient search procedure which allows for real-time implementation The authors describe an efficient procedure for searching such a large codebook deploying a focused search strategy, where less than 01% of the codebook is searched with performance very close to that of a full search High-quality speech at a bit rate of 13 kbps was obtained >

16 kbps wideband speech coding technique based on algebraic CELP

This paper describes the application of transform coded excitation (TCX) coding to encoding wideband speech and audio signals in the bit rate range of 16 kbits/s to 32 kbits/s. The approach uses a combination of time domain (linear prediction; pitch prediction) and frequency domain (transform coding; dynamic bit allocation) techniques, and utilizes a synthesis model similar to that of linear prediction coders such as CELP. However, at the encoder, the high complexity analysis-by-synthesis technique is bypassed by directly quantizing the so-called target signal in the frequency domain. The innovative excitation is derived at the decoder by inverse filtering the quantized target signal. The algorithm is intended for applications whereby a large number of bits is available for the innovative excitation. The TCX algorithm is utilized to encode wideband speech and audio signals with a 50-7000 Hz bandwidth. Novel quantization procedures including inter-frame prediction in the frequency domain are proposed to encode the target signal. The proposed algorithm achieves very high quality for speech at 16 kbits/s, and for music at 24 kbits/s. >

High quality coding of wideband audio signals using transform coded excitation (TCX)

This paper describes the GSM enhanced full rate (EFR) speech codec that has been standardised for the GSM mobile communication system. The GSM EFR codec has been jointly developed by Nokia and University of Sherbrooke. It provides speech quality at least equivalent to that of a wireline telephony reference (32 kbit/s ADPCM). The EFR codec uses 12.2 kbit/s for speech coding and 10.6 kbit/s for error protection. Speech coding is based on the ACELP algorithm (algebraic code excited linear prediction). The codec provides substantial quality improvement compared to the existing GSM full rate and half rate codecs. The old GSM codecs lack wireline quality even in error-free channel conditions, while the EFR codec provides wireline quality not only for error-free conditions but also for the most typical error conditions. With the EFR codec, wireline quality is also sustained in the presence of background noise and in tandem connections (mobile to mobile calls).

GSM enhanced full rate speech codec

This article describes the ITU-T Recommendation G.729 Annex A (G.729A) for encoding speech signals at 8 kb/s with low complexity. G.729A is the standard speech coding algorithm for multimedia digital simultaneous voice and data (DSVD). G.729A is bitstream interoperable with G.729; that is, speech coded with G.729A can be decoded with G.729, and vice versa. Like G.729, it uses the conjugate-structure algebraic code excited linear prediction (CS-ACELP) algorithm with 10 ms frames. However, several algorithmic changes have been introduced which result in a 50 percent reduction in complexity. This article describes the algorithm introduced to achieve the low complexity goal while meeting the terms of reference. Subjective tests showed that the performance of G.729A is equivalent to both G.729 and G.726 at 32 kb/s in most operating conditions; however, it is slightly worse in the case of three tandems and in the presence of background noise. A breakdown of the complexities of both G.729 and G.729A is also given.

R. Salami

Papers

The adaptive multirate wideband speech codec (AMR-WB)

16 kbps wideband speech coding technique based on algebraic CELP

High quality coding of wideband audio signals using transform coded excitation (TCX)

GSM enhanced full rate speech codec

ITU-T G.729 Annex A: reduced complexity 8 kb/s CS-ACELP codec for digital simultaneous voice and data