Abstract. In this paper we present first experimental results with a novel audio coding technique based on approximating Hilbert envelopes of relatively long segments of audio signal in critical-band-sized subbands by autoregressive model. We exploit the generalized autocorrelation linear predictive technique that allows for a better control of fitting the peaks and troughs of the envelope in the sub-band. Despite introducing longer algorithmic delay, improved coding efficiency is achieved. Since the described technique does not directly model short-term spectral envelopes of the signal, it is suitable not only for coding speech but also for coding of other audio signals.

In this paper we describe the first phase of development of our speech-to-speech system between English and Modem Persian under the DARPA Babylon program. We give an overview of the various system components: the front end ASR, the machine translation system and the speech generation system. Challenges such as the sparseness of available spoken language data and solutions that have been employed to maximize the obtained benefits from using these limited resources are examined.Efforts in the creation of the user interface and the underlying dialog management system for mediated communication are described.

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