SVMs for Automatic Speech Recognition: a Survey

Hidden Markov Models (HMMs) are, undoubtedly, the most employed core technique for Automatic Speech Recognition (ASR). Nevertheless, we are still far from achieving high-performance ASR systems. Some alternative approaches, most of them based on Artificial Neural Networks (ANNs), were proposed during the late eighties and early nineties. Some of them tackled the ASR problem using predictive ANNs, while others proposed hybrid HMM/ANN systems. However, despite some achievements, nowadays, the preponderance of Markov Models is a fact. During the last decade, however, a new tool appeared in the field of machine learning that has proved to be able to cope with hard classification problems in several fields of application: the Support Vector Machines (SVMs). The SVMs are effective discriminative classifiers with several outstanding characteristics, namely: their solution is that with maximum margin; they are capable to deal with samples of a very higher dimensionality; and their convergence to the minimum of the associated cost function is guaranteed. These characteristics have made SVMs very popular and successful. In this chapter we discuss their strengths and weakness in the ASR context and make a review of the current state-of-the-art techniques. We organize the contributions in two parts: isolated-word recognition and continuous speech recognition. Within the first part we review several techniques to produce the fixed-dimension vectors needed for original SVMs. Afterwards we explore more sophisticated techniques based on the use of kernels capable to deal with sequences of different length. Among them is the DTAK kernel, simple and effective, which rescues an old technique of speech recognition: Dynamic Time Warping (DTW). Within the second part, we describe some recent approaches to tackle more complex tasks like connected digit recognition or continuous speech recognition using SVMs. Finally we draw some conclusions and outline several ongoing lines of research

FPGA-Based Low-Power Speech Recognition with Recurrent Neural Networks

In this paper, a neural network based real-time speech recognition (SR) system is developed using an FPGA for very low-power operation. The implemented system employs two recurrent neural networks (RNNs); one is a speech-to-character RNN for acoustic modeling (AM) and the other is for character-level language modeling (LM). The system also employs a statistical word-level LM to improve the recognition accuracy. The results of the AM, the character-level LM, and the word-level LM are combined using a fairly simple N-best search algorithm instead of the hidden Markov model (HMM) based network. The RNNs are implemented using massively parallel processing elements (PEs) for low latency and high throughput. The weights are quantized to 6 bits to store all of them in the on-chip memory of an FPGA. The proposed algorithm is implemented on a Xilinx XC7Z045, and the system can operate much faster than real-time.Comment: Accepted to SiPS 201

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