207 research outputs found
Automatic Speech Recognition Using LP-DCTC/DCS Analysis Followed by Morphological Filtering
Front-end feature extraction techniques have long been a critical component in Automatic Speech Recognition (ASR). Nonlinear filtering techniques are becoming increasingly important in this application, and are often better than linear filters at removing noise without distorting speech features. However, design and analysis of nonlinear filters are more difficult than for linear filters. Mathematical morphology, which creates filters based on shape and size characteristics, is a design structure for nonlinear filters. These filters are limited to minimum and maximum operations that introduce a deterministic bias into filtered signals.
This work develops filtering structures based on a mathematical morphology that utilizes the bias while emphasizing spectral peaks. The combination of peak emphasis via LP analysis with morphological filtering results in more noise robust speech recognition rates.
To help understand the behavior of these pre-processing techniques the deterministic and statistical properties of the morphological filters are compared to the properties of feature extraction techniques that do not employ such algorithms. The robust behavior of these algorithms for automatic speech recognition in the presence of rapidly fluctuating speech signals with additive and convolutional noise is illustrated. Examples of these nonlinear feature extraction techniques are given using the Aurora 2.0 and Aurora 3.0 databases. Features are computed using LP analysis alone to emphasize peaks, morphological filtering alone, or a combination of the two approaches. Although absolute best results are normally obtained using a combination of the two methods, morphological filtering alone is nearly as effective and much more computationally efficient
Monaural speech separation using source-adapted models
We propose a model-based source separation system for use on single channel speech mixtures where the precise source characteristics are not known a priori. We do this by representing the space of source variation with a parametric signal model based on the eigenvoice technique for rapid speaker adaptation. We present an algorithm to infer the characteristics of the sources present in a mixture, allowing for significantly improved separation performance over that obtained using unadapted source models. The algorithm is evaluated on the task defined in the 2006 Speech Separation Challenge [1] and compared with separation using source-dependent models
A motion-based approach for audio-visual automatic speech recognition
The research work presented in this thesis introduces novel approaches for both visual
region of interest extraction and visual feature extraction for use in audio-visual
automatic speech recognition. In particular, the speakerâs movement that occurs
during speech is used to isolate the mouth region in video sequences and motionbased
features obtained from this region are used to provide new visual features for
audio-visual automatic speech recognition. The mouth region extraction approach
proposed in this work is shown to give superior performance compared with existing
colour-based lip segmentation methods. The new features are obtained from three
separate representations of motion in the region of interest, namely the difference in
luminance between successive images, block matching based motion vectors and
optical flow. The new visual features are found to improve visual-only and audiovisual
speech recognition performance when compared with the commonly-used
appearance feature-based methods.
In addition, a novel approach is proposed for visual feature extraction from either the
discrete cosine transform or discrete wavelet transform representations of the mouth
region of the speaker. In this work, the image transform is explored from a new
viewpoint of data discrimination; in contrast to the more conventional data
preservation viewpoint. The main findings of this work are that audio-visual
automatic speech recognition systems using the new features extracted from the
frequency bands selected according to their discriminatory abilities generally
outperform those using features designed for data preservation.
To establish the noise robustness of the new features proposed in this work, their
performance has been studied in presence of a range of different types of noise and at
various signal-to-noise ratios. In these experiments, the audio-visual automatic speech
recognition systems based on the new approaches were found to give superior
performance both to audio-visual systems using appearance based features and to
audio-only speech recognition systems
Speech Recognition
Chapters in the first part of the book cover all the essential speech processing techniques for building robust, automatic speech recognition systems: the representation for speech signals and the methods for speech-features extraction, acoustic and language modeling, efficient algorithms for searching the hypothesis space, and multimodal approaches to speech recognition. The last part of the book is devoted to other speech processing applications that can use the information from automatic speech recognition for speaker identification and tracking, for prosody modeling in emotion-detection systems and in other speech processing applications that are able to operate in real-world environments, like mobile communication services and smart homes
Automatic Recognition of Concurrent and Coupled Human Motion Sequences
We developed methods and algorithms for all parts of a motion recognition system, i. e. Feature Extraction, Motion Segmentation and Labeling, Motion Primitive and Context Modeling as well as Decoding. We collected several datasets to compare our proposed methods with the state-of-the-art in human motion recognition. The main contributions of this thesis are a structured functional motion decomposition and a flexible and scalable motion recognition system suitable for a Humanoid Robot
On adaptive decision rules and decision parameter adaptation for automatic speech recognition
Recent advances in automatic speech recognition are accomplished by designing a plug-in maximum a posteriori decision rule such that the forms of the acoustic and language model distributions are specified and the parameters of the assumed distributions are estimated from a collection of speech and language training corpora. Maximum-likelihood point estimation is by far the most prevailing training method. However, due to the problems of unknown speech distributions, sparse training data, high spectral and temporal variabilities in speech, and possible mismatch between training and testing conditions, a dynamic training strategy is needed. To cope with the changing speakers and speaking conditions in real operational conditions for high-performance speech recognition, such paradigms incorporate a small amount of speaker and environment specific adaptation data into the training process. Bayesian adaptive learning is an optimal way to combine prior knowledge in an existing collection of general models with a new set of condition-specific adaptation data. In this paper, the mathematical framework for Bayesian adaptation of acoustic and language model parameters is first described. Maximum a posteriori point estimation is then developed for hidden Markov models and a number of useful parameters densities commonly used in automatic speech recognition and natural language processing.published_or_final_versio
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