Dialogue Act Modeling for Automatic Tagging and Recognition of Conversational Speech

We describe a statistical approach for modeling dialogue acts in conversational speech, i.e., speech-act-like units such as Statement, Question, Backchannel, Agreement, Disagreement, and Apology. Our model detects and predicts dialogue acts based on lexical, collocational, and prosodic cues, as well as on the discourse coherence of the dialogue act sequence. The dialogue model is based on treating the discourse structure of a conversation as a hidden Markov model and the individual dialogue acts as observations emanating from the model states. Constraints on the likely sequence of dialogue acts are modeled via a dialogue act n-gram. The statistical dialogue grammar is combined with word n-grams, decision trees, and neural networks modeling the idiosyncratic lexical and prosodic manifestations of each dialogue act. We develop a probabilistic integration of speech recognition with dialogue modeling, to improve both speech recognition and dialogue act classification accuracy. Models are trained and evaluated using a large hand-labeled database of 1,155 conversations from the Switchboard corpus of spontaneous human-to-human telephone speech. We achieved good dialogue act labeling accuracy (65% based on errorful, automatically recognized words and prosody, and 71% based on word transcripts, compared to a chance baseline accuracy of 35% and human accuracy of 84%) and a small reduction in word recognition error.Comment: 35 pages, 5 figures. Changes in copy editing (note title spelling changed

Low cost duration modelling for noise robust speech recognition

State transition matrices as used in standard HMM decoders have two widely perceived limitations. One is that the implicit Geometric state duration distributions which they model do not accurately reflect true duration distributions. The other is that they impose no hard limit on maximum duration with the result that state transition probabilities often have little influence when combined with acoustic probabilities, which are of a different order of magnitude. Explicit duration models were developed in the past to address the first problem. These were not widely taken up because their performance advantage in clean speech recognition was often not sufficiently great to offset the extra complexity which they introduced. However, duration models have much greater potential when applied to noisy speech recognition. In this paper we present a simple and generic form of explicit duration model and show that this leads to strong performance improvements when applied to connected digit recognition in noise

Auxiliary Variables in Conditional Gaussian Mixtures for Automatic Speech Recognition

In previous work, we presented a case study using an estimated pitch value as the conditioning variable in conditional Gaussians that showed the utility of hiding the pitch values in certain situations or in modeling it independently of the hidden state in others. Since only single conditional Gaussians were used in that work, we extend that work here to using conditional Gaussian mixtures in the emission distributions to make this work more comparable to state-of-the-art automatic speech recognition. We also introduce a rate-of-speech (ROS) variable within the conditional Gaussian mixtures. We find that, under the current methods, using observed pitch or ROS in the recognition phase does not provide improvement. However, systems trained on pitch or ROS may provide improvement in the recognition phase over the baseline when the pitch or ROS is marginalized out

Modelling the effects of speech rate variation for automatic speech recognition

Wrede B. Modelling the effects of speech rate variation for automatic speech recognition. Bielefeld (Germany): Bielefeld University; 2002.In automatic speech recognition it is a widely observed phenomenon that variations in speech rate cause severe degradations of the speech recognition performance. This is due to the fact that standard stochastic based speech recognition systems specialise on average speech rate. Although many approaches to modelling speech rate variation have been made, an integrated approach in a substantial system still has be to developed. General approaches to rate modelling are based on rate dependent models which are trained with rate specific subsets of the training data. During decoding a signal based rate estimation is performed according to which the set of rate dependent models is selected. While such approaches are able to reduce the word error rate significantly, they suffer from shortcomings such as the reduction of training data and the expensive training and decoding procedure. However, phonetic investigations show that there is a systematic relationship between speech rate and the acoustic characteristics of speech. In fast speech a tendency of reduction can be observed which can be described in more detail as a centralisation effect and an increase in coarticulation. Centralisation means that the formant frequencies of vowels tend to shift towards the vowel space center while increased coarticulation denotes the tendency of the spectral features of a vowel to shift towards those of its phonemic neighbour. The goal of this work is to investigate the possibility to incorporate the knowledge of the systematic nature of the influence of speech rate variation on the acoustic features in speech rate modelling. In an acoustic-phonetic analysis of a large corpus of spontaneous speech it was shown that an increased degree of the two effects of centralisation and coarticulation can be found in fast speech. Several measures for these effects were developed and used in speech recognition experiments with rate dependent models. A thorough investigation of rate dependent models showed that with duration and coarticulation based measures significant increases of the performance could be achieved. It was shown that by the use of different measures the models were adapted either to centralisation or coarticulation. Further experiments showed that by a more detailed modelling with more rate classes a further improvement can be achieved. It was also observed that a general basis for the models is needed before rate adaptation can be performed. In a comparison to other sources of acoustic variation it was shown that the effects of speech rate are as severe as those of speaker variation and environmental noise. All these results show that for a more substantial system that models rate variations accurately it is necessary to focus on both, durational and spectral effects. The systematic nature of the effects indicates that a continuous modelling is possible

Using duration information in HMM-based automatic speech recognition.

Zhu Yu.Thesis (M.Phil.)--Chinese University of Hong Kong, 2005.Includes bibliographical references (leaves 100-104).Abstracts in English and Chinese.Chapter CHAPTER 1 --- lNTRODUCTION --- p.1Chapter 1.1. --- Speech and its temporal structure --- p.1Chapter 1.2. --- Previous work on the modeling of temporal structure --- p.1Chapter 1.3. --- Integrating explicit duration modeling in HMM-based ASR system --- p.3Chapter 1.4. --- Thesis outline --- p.3Chapter CHAPTER 2 --- BACKGROUND --- p.5Chapter 2.1. --- Automatic speech recognition process --- p.5Chapter 2.2. --- HMM for ASR --- p.6Chapter 2.2.1. --- HMM for ASR --- p.6Chapter 2.2.2. --- HMM-based ASR system --- p.7Chapter 2.3. --- General approaches to explicit duration modeling --- p.12Chapter 2.3.1. --- Explicit duration modeling --- p.13Chapter 2.3.2. --- Training of duration model --- p.16Chapter 2.3.3. --- Incorporation of duration model in decoding --- p.18Chapter CHAPTER 3 --- CANTONESE CONNECTD-DlGlT RECOGNITION --- p.21Chapter 3.1. --- Cantonese connected digit recognition --- p.21Chapter 3.1.1. --- Phonetics of Cantonese and Cantonese digit --- p.21Chapter 3.2. --- The baseline system --- p.24Chapter 3.2.1. --- Speech corpus --- p.24Chapter 3.2.2. --- Feature extraction --- p.25Chapter 3.2.3. --- HMM models --- p.26Chapter 3.2.4. --- HMM decoding --- p.27Chapter 3.3. --- Baseline performance and error analysis --- p.27Chapter 3.3.1. --- Recognition performance --- p.27Chapter 3.3.2. --- Performance for different speaking rates --- p.28Chapter 3.3.3. --- Confusion matrix --- p.30Chapter CHAPTER 4 --- DURATION MODELING FOR CANTONESE DIGITS --- p.41Chapter 4.1. --- Duration features --- p.41Chapter 4.1.1. --- Absolute duration feature --- p.41Chapter 4.1.2. --- Relative duration feature --- p.44Chapter 4.2. --- Parametric distribution for duration modeling --- p.47Chapter 4.3. --- Estimation of the model parameters --- p.51Chapter 4.4. --- Speaking-rate-dependent duration model --- p.52Chapter CHAPTER 5 --- USING DURATION MODELING FOR CANTONSE DIGIT RECOGNITION --- p.57Chapter 5.1. --- Baseline decoder --- p.57Chapter 5.2. --- Incorporation of state-level duration model --- p.59Chapter 5.3. --- Incorporation word-level duration model --- p.62Chapter 5.4. --- Weighted use of duration model --- p.65Chapter CHAPTER 6 --- EXPERIMENT RESULT AND ANALYSIS --- p.66Chapter 6.1. --- Experiments with speaking-rate-independent duration models --- p.66Chapter 6.1.1. --- Discussion --- p.68Chapter 6.1.2. --- Analysis of the error patterns --- p.71Chapter 6.1.3. --- "Reduction of deletion, substitution and insertion" --- p.72Chapter 6.1.4. --- Recognition performance at different speaking rates --- p.75Chapter 6.2. --- Experiments with speaking-rate-dependent duration models --- p.77Chapter 6.2.1. --- Using true speaking rate --- p.77Chapter 6.2.2. --- Using estimated speaking rate --- p.79Chapter 6.3. --- Evaluation on another speech database --- p.80Chapter 6.3.1. --- Experimental setup --- p.80Chapter 6.3.2. --- Experiment results and analysis --- p.82Chapter CHAPTER 7 --- CONCLUSIONS AND FUTUR WORK --- p.87Chapter 7.1. --- Conclusion and understanding of current work --- p.87Chapter 7.2. --- Future work --- p.89Chapter A --- APPENDIX --- p.90BIBLIOGRAPHY --- p.10

Speech recognition with auxiliary information

Automatic speech recognition (ASR) is a very challenging problem due to the wide variety of the data that it must be able to deal with. Being the standard tool for ASR, hidden Markov models (HMMs) have proven to work well for ASR when there are controls over the variety of the data. Being relatively new to ASR, dynamic Bayesian networks (DBNs) are more generic models with algorithms that are more flexible than those of HMMs. Various assumptions can be changed without modifying the underlying algorithm and code, unlike in HMMs; these assumptions relate to the variables to be modeled, the statistical dependencies between these variables, and the observations which are available for certain of the variables. The main objective of this thesis, therefore, is to examine some areas where DBNs can be used to change HMMs' assumptions so as to have models that are more robust to the variety of data that ASR must deal with. HMMs model the standard observed features by jointly modeling them with a hidden discrete state variable and by having certain restraints placed upon the states and features. Some of the areas where DBNs can generalize this modeling framework of HMMs involve the incorporation of even more "auxiliary" variables to help the modeling which HMMs typically can only do with the two variables under certain restraints. The DBN framework is more flexible in how this auxiliary variable is introduced in different ways. First, this auxiliary information aids the modeling due to its correlation with the standard features. As such, in the DBN framework, we can make it directly condition the distribution of the standard features. Second, some types of auxiliary information are not strongly correlated with the hidden state. So, in the DBN framework we may want to consider the auxiliary variable to be conditionally independent of the hidden state variable. Third, as auxiliary information tends to be strongly correlated with its previous values in time, I show DBNs using discretized auxiliary variables that model the evolution of the auxiliary information over time. Finally, as auxiliary information can be missing or noisy in using a trained system, the DBNs can do recognition using just its prior distribution, learned on auxiliary information observations during training. I investigate these different advantages of DBN-based ASR using auxiliary information involving articulator positions, estimated pitch, estimated rate-of-speech, and energy. I also show DBNs to be better at incorporating auxiliary information than hybrid HMM/ANN ASR, using artificial neural networks (ANNs). I show how auxiliary information is best introduced in a time-dependent manner. Finally, DBNs with auxiliary information are better able than standard HMM approaches to handling noisy speech; specifically, DBNs with hidden energy as auxiliary information -- that conditions the distribution of the standard features and which is conditionally independent of the state -- are more robust to noisy speech than HMMs are

Using auxiliary sources of knowledge for automatic speech recognition

Standard hidden Markov model (HMM) based automatic speech recognition (ASR) systems usually use cepstral features as acoustic observation and phonemes as subword units. Speech signal exhibits wide range of variability such as, due to environmental variation, speaker variation. This leads to different kinds of mismatch, such as, mismatch between acoustic features and acoustic models or mismatch between acoustic features and pronunciation models (given the acoustic models). The main focus of this work is on integrating auxiliary knowledge sources into standard ASR systems so as to make the acoustic models more robust to the variabilities in the speech signal. We refer to the sources of knowledge that are able to provide additional information about the sources of variability as auxiliary sources of knowledge. The auxiliary knowledge sources that have been primarily investigated in the present work are auxiliary features and auxiliary subword units. Auxiliary features are secondary source of information that are outside of the standard cepstral features. They can be estimation from the speech signal (e.g., pitch frequency, short-term energy and rate-of-speech), or additional measurements (e.g., articulator positions or visual information). They are correlated to the standard acoustic features, and thus can aid in estimating better acoustic models, which would be more robust to variabilities present in the speech signal. The auxiliary features that have been investigated are pitch frequency, short-term energy and rate-of-speech. These features can be modelled in standard ASR either by concatenating them to the standard acoustic feature vectors or by using them to condition the emission distribution (as done in gender-based acoustic modelling). We have studied these two approaches within the framework of hybrid HMM/artificial neural networks based ASR, dynamic Bayesian network based ASR and TANDEM system on different ASR tasks. Our studies show that by modelling auxiliary features along with standard acoustic features the performance of the ASR system can be improved in both clean and noisy conditions. We have also proposed an approach to evaluate the adequacy of the baseform pronunciation model of words. This approach allows us to compare between different acoustic models as well as to extract pronunciation variants. Through the proposed approach to evaluate baseform pronunciation model, we show that the matching and discriminative properties of single baseform pronunciation can be improved by integrating auxiliary knowledge sources in standard ASR. Standard ASR systems use usually phonemes as the subword units in a Markov chain to model words. In the present thesis, we also study a system where word models are described by two parallel chains of subword units: one for phonemes and the other are for graphemes (phoneme-grapheme based ASR). Models for both types of subword units are jointly learned using maximum likelihood training. During recognition, decoding is performed using either or both of the subword unit chains. In doing so, we thus have used graphemes as auxiliary subword units. The main advantage of using graphemes is that the word models can be defined easily using the orthographic transcription, thus being relatively noise free as compared to word models based upon phoneme units. At the same time, there are drawbacks to using graphemes as subword units, since there is a weak correspondence between the grapheme and the phoneme in languages such as English. Experimental studies conducted for American English on different ASR tasks have shown that the proposed phoneme-grapheme based ASR system can perform better than the standard ASR system that uses only phonemes as its subword units. Furthermore, while modelling context-dependent graphemes (similar to context-dependent phonemes), we observed that context-dependent graphemes behave like phonemes. ASR studies conducted on different tasks showed that by modelling context-dependent graphemes only (without any phonetic information) performance competitive to the state-of-the-art context-dependent phoneme-based ASR system can be obtained

Speech Recognition Using On-Line Estimation Of Speaking Rate

In this paper, we describe a rate of speech estimator that is derived directly from the acoustic signal. This measure has been developed as an alternative to lexical measures of speaking rate such as phones or syllables per second, which, in previous work, we estimated using a first recognition pass; the accuracy of our earlier lexical rate estimate depended on the quality of recognition. Here we show that our new measure is a good predictor of word error rate, and in addition, correlates moderately well with lexical speech rate. We also show that a simple modification of the model transition probabilities based on this measure can reduce the error rate almost as much as using lexical phones per second calculated from manually transcribed data. When we categorized test utterances based on speaking rate thresholds computed from the training set, we observed that a different transition probability value was required to minimize the error rate in each speaking rate bin. However, the reduc..