Recognizing GSM Digital Speech

The Global System for Mobile (GSM) environment encompasses three main problems for automatic speech recognition (ASR) systems: noisy scenarios, source coding distortion, and transmission errors. The first one has already received much attention; however, source coding distortion and transmission errors must be explicitly addressed. In this paper, we propose an alternative front-end for speech recognition over GSM networks. This front-end is specially conceived to be effective against source coding distortion and transmission errors. Specifically, we suggest extracting the recognition feature vectors directly from the encoded speech (i.e., the bitstream) instead of decoding it and subsequently extracting the feature vectors. This approach offers two significant advantages. First, the recognition system is only affected by the quantization distortion of the spectral envelope. Thus, we are avoiding the influence of other sources of distortion as a result of the encoding-decoding process. Second, when transmission errors occur, our front-end becomes more effective since it is not affected by errors in bits allocated to the excitation signal. We have considered the half and the full-rate standard codecs and compared the proposed front-end with the conventional approach in two ASR tasks, namely, speaker-independent isolated digit recognition and speaker-independent continuous speech recognition. In general, our approach outperforms the conventional procedure, for a variety of simulated channel conditions. Furthermore, the disparity increases as the network conditions worsen

Modelling the effects of spontaneous speech in speech recognition

Intrinsic variability of the speaker in spontaneous speech remains a challenge to state of the art Automatic speech recognition (ASR). While planned speech exhibits a moderate variability, the significant variability of spontaneous speech is caused by situation, context, intention, emotion and listeners. This conditioning of speech is observable in terms of speaking rate and in feature space. We analysed broadcast news (BN) and broadcast conversational (BC) speech in terms of phoneme rate (PR) and feature space reduction (FSR), and contrasted both with the planned speech data. Strong statistically significant differences were revealed. We cluster the speech segments with respect to their degree of PR and FSR forming a set of variability classes, and induce the variability classes into the Hidden-Markov-Model (HMM) based acoustic model (AM). In recognition we follow two approaches: the first considers the variability class as context variable, the second relies on prior estimation of the variability class after the first pass of a multi-pass recognition system. Beside explicit modelling of the intrinsic speech variability of the speaker, we furthermore segregate the general speaker specific characteristics by means of speaker adaptive training (SAT) into feature space transforms using ConstrainedMaximumLikelihood Linear Regression (CMLLR), and apply the adaptive approach in third pass recognition. By approaching to model both within speaker variation and between speaker variation in spontaneous speech, we address two fundamental sources of speech variability that determine the performance of ASR systems.Peer ReviewedPostprint (published version