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A concept and architecture of a personal communication system (PCS) is introduced that integrates audio communication and hearing support for the elderly and hearing-impaired through a personal hearing system (PHS). The concept envisions a central processor connected to audio headsets via a wireless body area network (WBAN). To demonstrate the concept, a prototype PCS is presented that is implemented on a netbook computer with a dedicated audio interface in combination with a mobile phone. The prototype can be used for field-testing possible applications and to reveal possibilities and limitations of the concept of integrating hearing support in consumer audio communication devices. It is shown that the prototype PCS can integrate hearing aid functionality, telephony, public announcement systems, and home entertainment. An exemplary binaural speech enhancement scheme that represents a large class of possible PHS processing schemes is shown to be compatible with the general concept. However, an analysis of hardware and software architectures shows that the implementation of a PCS on future advanced cell phone-like devices is challenging. Because of limitations in processing power, recoding of prototype implementations into fixed point arithmetic will be required and WBAN performance is still a limiting factor in terms of data rate and delay.

This paper describes the narrow-band speech coder that was proposed by France as a candidate to the new NATO standard STANAG-4591. This dual-rate coder at 1.2 and 2.4 kbps is based on the HSX (Harmonic & Stochastic eXcitation) algorithm.. The 2.4 kbps version was first developed by the University of Sherbrooke [1], then refined by Thales Communications (formerly Thomson-CSF Communications). An extension at 1.2 kbps was later developed by Thales Communications. This half-rate version uses the same algorithmic core than the 2.4 kbps version, but achieves a lower bit rate by grouping 3 consecutive frames grouped into a single 67.5 ms super-frame [2]. A noise reduction procedure is considered as a part of the next NATO low bit rate speech coding standard.. This noise reduction system is based on the combination of two different noise reduction techniques. The first one is the Wiener filter under signal presence uncertainty. The second one is Lim's filter, that takes into account the auto-regressive model of human voice [3]. These new coders provide a low complexity solution with increased intelligibility and quality compatible with a growing number of military, professional and commercial applications.

This paper describes a prospective study of the contribution of a single-sensor noise pre-processing method, prior to coding, to the performance of a parametric low bit rate speech coder in adverse conditions. The 2.4kbits/s vocoder we use estimates four parameters: fundamental frequency, voicing, linear prediction coefficients and energy. Firstly, we study the influence of different noise levels on the estimated parameters with and without noise reduction system. Secondly, we measure the contribution of (i) each speech coder parameter and (ii) the speech enhancement system to the global output intelligibility. Finally, results show the interest of such a speech enhancement system for low bit rate parameter estimation and underline the interest to adapt different pre-processing techniques for each parameter estimation.