Human and computer recognition of regional accents and ethnic groups from British English speech

The variability of the channel and environment is one of the most important factors affecting the performance of text-independent speaker verification systems. The best techniques for channel compensation are model based. Most of them have been proposed for Gaussian mixture models, while in the feature domain blind channel compensation is usually performed. The aim of this work is to explore techniques that allow more accurate intersession compensation in the feature domain. Compensating the features rather than the models has the advantage that the transformed parameters can be used with models of a different nature and complexity and for different tasks. In this paper, we evaluate the effects of the compensation of the intersession variability obtained by means of the channel factors approach. In particular, we compare channel variability modeling in the usual Gaussian mixture model domain, and our proposed feature domain compensation technique. We show that the two approaches lead to similar results on the NIST 2005 Speaker Recognition Evaluation data with a reduced computation cost. We also report the results of a system, based on the intersession compensation technique in the feature space that was among the best participants in the NIST 2006 Speaker Recognition Evaluation. Moreover, we show how we obtained significant performance improvement in language recognition by estimating and compensating, in the feature domain, the distortions due to interspeaker variability within the same language.

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Compensation of Nuisance Factors for Speaker and Language Recognition

A fundamental challenge for current research on speech science and technology is understanding and modeling individual variation in spoken language. Individuals have their own speaking styles, depending on many factors, such as their dialect and accent as well as their socioeconomic background. These individual differences typically introduce modeling difficulties for large-scale speaker-independent systems designed to process input from any variant of a given language. This dissertation focuses on automatically identifying the dialect or accent of a speaker given a sample of their speech, and demonstrates how such a technology can be employed to improve Automatic Speech Recognition (ASR).
In this thesis, we describe a variety of approaches that make use of multiple streams of information in the acoustic signal to build a system that recognizes the regional dialect and accent of a speaker. In particular, we examine frame-based acoustic, phonetic, and phonotactic features, as well as high-level prosodic features, comparing generative and discriminative modeling techniques. We first analyze the effectiveness of approaches to language identification that have been successfully employed by that community, applying them here to dialect identification. We next show how we can improve upon these techniques. Finally, we introduce several novel modeling approaches – Discriminative Phonotactics and kernel-based methods. We test our best performing approach on four broad Arabic dialects, ten Arabic sub-dialects, American English vs. Indian English accents, American English Southern vs. Non-Southern, American dialects at the state level plus Canada, and three Portuguese dialects.
Our experiments demonstrate that our novel approach, which relies on the hypothesis that certain phones are realized differently across dialects, achieves new state-of-the-art performance on most dialect recognition tasks. This approach achieves an Equal Error Rate (EER) of 4% for four broad Arabic dialects, an EER of 6.3% for American vs. Indian English accents, 14.6% for American English Southern vs. Non-Southern dialects, and 7.9% for three Portuguese dialects. Our framework can also be used to automatically extract linguistic knowledge, specifically the context-dependent phonetic cues that may distinguish one dialect form another. We illustrate the efficacy of our approach by demonstrating the correlation of our results with geographical proximity of the various dialects.
As a final measure of the utility of our studies, we also show that, it is possible to improve ASR. Employing our dialect identification system prior to ASR to identify the Levantine Arabic dialect in mixed speech of a variety of dialects allows us to optimize the engine's language model and use Levantine-specific acoustic models where appropriate. This procedure improves the Word Error Rate (WER) for Levantine by 4.6% absolute; 9.3% relative. In addition, we demonstrate in this thesis that, using a linguistically-motivated pronunciation modeling approach, we can improve the WER of a state-of-the art ASR system by 2.2% absolute and 11.5% relative WER on Modern Standard Arabic.

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Automatic dialect and accent recognition and its application to speech recognition

This paper presents a description of the MIT Lincoln Laboratory language recognition system submitted to the NIST 2009 Language Recognition Evaluation (LRE). This system consists of a fusion of three core recognizers, two based on spectral similarity and one based on tokenization. The 2009 LRE differed from previous ones in that test data included narrowband segments from worldwide Voice of America broadcasts as well as conventional recorded conversational telephone speech. Results are presented for the 23-language closed-set and open-set detection tasks at the 30, 10, and 3 second durations along with a discussion of the language-pair task. On the 30 second 23-language closed set detection task, the system achieved a 1.64 average error rate.

The MITLL NIST LRE 2009 language recognition system

Conversations in the intelligent vehicles are usually short utterance. As the durations of the short utterances are small (e.g., less than 3 s), it is difficult to learn sufficient information to distinguish the type of languages. In this paper, we propose an end-to-end short utterances based speech language identification (SLI) approach, which is especially suitable for the short utterance based language identification. This approach is implemented with a long short-term memory (LSTM) neural network, which is designed for the SLI application in intelligent vehicles. The features used for LSTM learning are generated by a transfer learning method. The bottleneck features of a deep neural network, which are obtained for a mandarin acoustic-phonetic classifier, are used for the LSTM training. In order to improve the SLD accuracy with short utterances, a phase vocoder based time-scale modification method is utilized to reduce/increase the speech rate of the test utterance. By connecting the normal, speech rate reduced, and speech rate increased utterances, we can extend the length of the test utterances such that the performance of the SLI system is improved. The experimental results on the AP17-OLR database demonstrate that the proposed method can improve the performance of SLD, especially on short utterance. The proposed SLI has robust performance under the vehicular noisy environment.

Short Utterance Based Speech Language Identification in Intelligent Vehicles With Time-Scale Modifications and Deep Bottleneck Features

This paper presents the language identification (LID) system developed in Speech@FIT group at Brno University of Technology (BUT) for NIST 2005 Language Recognition Evaluation. The system consists of two parts: phonotactic and acoustic. Phonotactic system is based on hybrid phoneme recognizers trained on SpeechDat-E database. Phoneme lattices are used to train and test phonotactic language models. Further improvement is obtained by using anti-models. Acoustic system is based on GMM modeling trained under Maximum Mutual Information framework. We describe both parts and provide a discussion of performance on LRE 2005 recognition task.

P. Sckwarz

Papers

Brno University of Technology System for NIST 2005 Language Recognition Evaluation