摘要:
A multiple confidence measures subsystem of an automated speech recognition system allows otherwise independent confidence measures to be integrated and used for both training and testing on a consistent basis. Speech to be recognized is input to a speech recognizer and a recognition verifier of the multiple confidence measures subsystem. The speech recognizer generates one or more confidence measures. The speech recognizer preferably generates a misclassification error (MCE) distance as one of the confidence measures. The recognized speech output by the speech recognizer is input to the recognition verifier, which outputs one or more confidence measures. The recognition verifier preferably outputs a misverification error (MVE) distance as one of the confidence measures. The confidence measures output by the speech recognizer and the recognition verifier are normalized and then input to an integrator. The integrator integrates the various confidence measures during both a training phase for the hidden Markov models implemented in the speech recognizer and the recognition verifier and during testing of the input speech. The integrator is preferably implemented using a multi-layer perceptron (MLP). The output of the integrator, rather than the recognition verifier, determines whether the recognized utterance hypothesis generated by the speech recognizer should be accepted or rejected.
摘要:
An unsupervised, discriminative, sentence level, HMM adaptation based on speech-silence classification is presented. Silence and speech regions are determined either using a speech end-pointer or the segmentation obtained from the recognizer in a first pass. The discriminative training procedure using a GPD or any other discriminative training algorithm, employed in conjunction with the HMM-based recognizer, is then used to increase the discrimination between silence and speech.
摘要:
A speech spurt detecting apparatus for detecting speech spurts in a voice signal has a storage for storing an input voice signal. A decision portion determines speech spurt sections and mute sections using a threshold value and sets one of the mute sections at a latter part of a hangover time. A mute level statistical processor estimates the noise distribution of a signal in the mute sections. A speech spurt detecting threshold value decision portion receives the average and the variance of the noise distribution from the mute level statistical processor and approximates the noise distribution to a gamma distribution to decide a speech spurt detecting threshold. A speech spurt transmitting portion outputs the voice signal in the speech spurt sections from the storage. A speech spurt level statistical processor carries out statistical processing of the speech spurt sections. The speech spurt detecting threshold value decision portion detects an error of the speech spurt detecting threshold value using the speech spurt level statistical processor and the mute level statistical processor and resets the speech spurt detecting threshold value to its initial value if the error exceeds a predetermined value. The speech spurt detecting threshold value decision portion increases the speech spurt detecting threshold value at a fixed rate in each of the speech spurt sections, and computes (the average).sup.2 /(the variance) to obtain an adjusting coefficient and computes (the adjusting coefficient).times.(the average) to obtain the speech spurt detecting threshold value.
摘要:
A hypertext navigation system that is controllable by spoken words have hypertext documents to which specific dictionaries and probability models for assisting in an acoustic voice recognition of hyper-links of this hypertext document are allocated. Control of a hypertext viewer or, respectively, browser and navigation in the hypertext document or hypertext system by pronouncing links is provided. The voice recognition is thereby optimally adapted to the links to be recognized without these having to be previously known.
摘要:
In a speaker recognition system, a tree-structured reference pattern storing unit has first through M-th node stages each of which has nodes that respectively store a reference pattern of inhibiting speakers. The reference pattern of each node of (N-1)-th node stage represents acoustic features in the reference patterns of predetermined ones of the nodes of the N-th node stage. An analysis unit analyzes input speech and converts the input speech into feature vectors. A similarities calculating unit calculates similarities between the feature vectors and the reference patterns of all of the inhibiting speakers. An inhibiting speaker selecting unit sorts the similarities and selects a predetermined number of inhibiting speakers. The similarities calculating unit calculates the similarity of the node of the first node stage and calculates the similarities of ones of the nodes of the N-th node stage which are connected to a predetermined number of nodes of the (N-1)-th node stage, selected in an order based on highest similarities.
摘要:
Systems and methods consistent with the present invention determine whether to accept one of a plurality of intermediate recognition results output by a speech recognition system as a final recognition result. The system first combines a plurality of speech rejection features into a feature function in which weights are assigned to each rejection feature in accordance with a recognition accuracy of each rejection feature. Feature values are then calculated for each of the rejection features using the plurality of intermediate recognition results. The system next computes the feature function according to the calculated feature values to determine a rejection decision value. Finally, one of the plurality of intermediate recognition results is accepted as the final recognition result according to the rejection decision value.
摘要:
The present invention is a computer software system that allows the developer of a speech-enabled system to create a grammar and corpus for use in the system. A table interface is used, and phrases in the grammar are entered into cells in the table. The table also includes token data which corresponds to each valid utterance. When the grammar is defined, the computer software system automatically traverses the table to enumerate all possible valid utterances in the grammar. This traversal generates a listing (corpus) of valid utterances and their respective tokens. This listing can then be used to interpret spoken utterances for a speech-enabled system. The computer software system also transcribes the grammar rules found in the table to a format compatible with a variety of supported commercially-available speech recognizers.
摘要:
The invention relates to a method and an apparatus for adding a new entry to a speech recognition dictionary, more particularly to a system and method for generating transcriptions from multiple utterances of a given word. The novel method and apparatus automatically transcribes several training utterances into transcriptions without knowledge of the orthography of the word being added. It also provides a method and apparatus for transcribing multiple utterances into a single transcription that can be added to a speech recognition dictionary. In a first step, each utterance is analyzed individually to get their respective acoustic characteristics. Following this, these characteristics are combined to generate a set of the most likely transcriptions using the acoustic information obtained from each of the training utterances.
摘要:
A method and system performs speech recognition training using Hidden Markov Models. Initially, preprocessed speech signals that include a plurality of observations are stored by the system. Initial Hidden Markov Model (HMM) parameters are then assigned. Summations are then calculated using modified equations derived substantially from the following equations, wherein u.ltoreq.v
摘要:
A speech recognition system which effectively recognizes unknown speech from multiple acoustic environments includes a set of secondary models, each associated with one or more particular acoustic environments, integrated with a base set of recognition models. The speech recognition system is trained by making a set of secondary models in a first stage of training, and integrating the set of secondary models with a base set of recognition models in a second stage of training.