Pub Date : 2013-12-01DOI: 10.1109/ASRU.2013.6707700
A. Sethy, Stanley F. Chen, E. Arisoy, B. Ramabhadran, Kartik Audhkhasi, Shrikanth S. Narayanan, Paul Vozila
For many speech recognition tasks, the best language model performance is achieved by collecting text from multiple sources or domains, and interpolating language models built separately on each individual corpus. When multiple corpora are available, it has also been shown that when using a domain adaptation technique such as feature augmentation [1], the performance on each individual domain can be improved by training a joint model across all of the corpora. In this paper, we explore whether improving each domain model via joint training also improves performance when interpolating the models together. We show that the diversity of the individual models is an important consideration, and propose a method for adjusting diversity to optimize overall performance. We present results using word n-gram models and Model M, a class-based n-gram model, and demonstrate improvements in both perplexity and word-error rate relative to state-of-the-art results on a Broadcast News transcription task.
{"title":"Joint training of interpolated exponential n-gram models","authors":"A. Sethy, Stanley F. Chen, E. Arisoy, B. Ramabhadran, Kartik Audhkhasi, Shrikanth S. Narayanan, Paul Vozila","doi":"10.1109/ASRU.2013.6707700","DOIUrl":"https://doi.org/10.1109/ASRU.2013.6707700","url":null,"abstract":"For many speech recognition tasks, the best language model performance is achieved by collecting text from multiple sources or domains, and interpolating language models built separately on each individual corpus. When multiple corpora are available, it has also been shown that when using a domain adaptation technique such as feature augmentation [1], the performance on each individual domain can be improved by training a joint model across all of the corpora. In this paper, we explore whether improving each domain model via joint training also improves performance when interpolating the models together. We show that the diversity of the individual models is an important consideration, and propose a method for adjusting diversity to optimize overall performance. We present results using word n-gram models and Model M, a class-based n-gram model, and demonstrate improvements in both perplexity and word-error rate relative to state-of-the-art results on a Broadcast News transcription task.","PeriodicalId":265258,"journal":{"name":"2013 IEEE Workshop on Automatic Speech Recognition and Understanding","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126690312","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2013-12-01DOI: 10.1109/ASRU.2013.6707728
S. Wegmann, Arlo Faria, Adam L. Janin, K. Riedhammer, N. Morgan
In this paper we apply diagnostic analysis to gain a deeper understanding of the performance of the the keyword search system that we have developed for conversational telephone speech in the IARPA Babel program. We summarize the Babel task, its primary performance metric, “actual term weighted value” (ATWV), and our recognition and keyword search systems. Our analysis uses two new oracle ATWV measures, a bootstrap-based ATWV confidence interval, and includes a study of the underpinnings of the large ATWV gains due to system combination. This analysis quantifies the potential ATWV gains from improving the number of true hits and the overall quality of the detection scores in our system's posting lists. It also shows that system combination improves our systems' ATWV via a small increase in the number of true hits in the posting lists.
{"title":"The TAO of ATWV: Probing the mysteries of keyword search performance","authors":"S. Wegmann, Arlo Faria, Adam L. Janin, K. Riedhammer, N. Morgan","doi":"10.1109/ASRU.2013.6707728","DOIUrl":"https://doi.org/10.1109/ASRU.2013.6707728","url":null,"abstract":"In this paper we apply diagnostic analysis to gain a deeper understanding of the performance of the the keyword search system that we have developed for conversational telephone speech in the IARPA Babel program. We summarize the Babel task, its primary performance metric, “actual term weighted value” (ATWV), and our recognition and keyword search systems. Our analysis uses two new oracle ATWV measures, a bootstrap-based ATWV confidence interval, and includes a study of the underpinnings of the large ATWV gains due to system combination. This analysis quantifies the potential ATWV gains from improving the number of true hits and the overall quality of the detection scores in our system's posting lists. It also shows that system combination improves our systems' ATWV via a small increase in the number of true hits in the posting lists.","PeriodicalId":265258,"journal":{"name":"2013 IEEE Workshop on Automatic Speech Recognition and Understanding","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132658268","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2013-12-01DOI: 10.1109/ASRU.2013.6707757
Raymond Brueckner, Björn Schuller
With the impressive advances of deep learning in recent years the interest in neural networks has resurged in the fields of automatic speech recognition and emotion recognition. In this paper we apply neural networks to address speaker-independent detection and classification of laughter and filler vocalizations in speech. We first explore modeling class posteriors with standard neural networks and deep stacked autoencoders. Then, we adopt a hierarchical neural architecture to compute enhanced class posteriors and demonstrate that this approach introduces significant and consistent improvements on the Social Signals Sub-Challenge of the Interspeech 2013 Computational Paralinguistics Challenge (ComParE). On this task we achieve a value of 92.4% of the unweighted average area-under-the-curve, which is the official competition measure, on the test set. This constitutes an improvement of 9.1% over the baseline and is the best result obtained so far on this task.
{"title":"Hierarchical neural networks and enhanced class posteriors for social signal classification","authors":"Raymond Brueckner, Björn Schuller","doi":"10.1109/ASRU.2013.6707757","DOIUrl":"https://doi.org/10.1109/ASRU.2013.6707757","url":null,"abstract":"With the impressive advances of deep learning in recent years the interest in neural networks has resurged in the fields of automatic speech recognition and emotion recognition. In this paper we apply neural networks to address speaker-independent detection and classification of laughter and filler vocalizations in speech. We first explore modeling class posteriors with standard neural networks and deep stacked autoencoders. Then, we adopt a hierarchical neural architecture to compute enhanced class posteriors and demonstrate that this approach introduces significant and consistent improvements on the Social Signals Sub-Challenge of the Interspeech 2013 Computational Paralinguistics Challenge (ComParE). On this task we achieve a value of 92.4% of the unweighted average area-under-the-curve, which is the official competition measure, on the test set. This constitutes an improvement of 9.1% over the baseline and is the best result obtained so far on this task.","PeriodicalId":265258,"journal":{"name":"2013 IEEE Workshop on Automatic Speech Recognition and Understanding","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130663355","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2013-12-01DOI: 10.1109/ASRU.2013.6707763
Yajie Miao, Florian Metze, Shourabh Rawat
As a feed-forward architecture, the recently proposed maxout networks integrate dropout naturally and show state-of-the-art results on various computer vision datasets. This paper investigates the application of deep maxout networks (DMNs) to large vocabulary continuous speech recognition (LVCSR) tasks. Our focus is on the particular advantage of DMNs under low-resource conditions with limited transcribed speech. We extend DMNs to hybrid and bottleneck feature systems, and explore optimal network structures (number of maxout layers, pooling strategy, etc) for both setups. On the newly released Babel corpus, behaviors of DMNs are extensively studied under different levels of data availability. Experiments show that DMNs improve low-resource speech recognition significantly. Moreover, DMNs introduce sparsity to their hidden activations and thus can act as sparse feature extractors.
{"title":"Deep maxout networks for low-resource speech recognition","authors":"Yajie Miao, Florian Metze, Shourabh Rawat","doi":"10.1109/ASRU.2013.6707763","DOIUrl":"https://doi.org/10.1109/ASRU.2013.6707763","url":null,"abstract":"As a feed-forward architecture, the recently proposed maxout networks integrate dropout naturally and show state-of-the-art results on various computer vision datasets. This paper investigates the application of deep maxout networks (DMNs) to large vocabulary continuous speech recognition (LVCSR) tasks. Our focus is on the particular advantage of DMNs under low-resource conditions with limited transcribed speech. We extend DMNs to hybrid and bottleneck feature systems, and explore optimal network structures (number of maxout layers, pooling strategy, etc) for both setups. On the newly released Babel corpus, behaviors of DMNs are extensively studied under different levels of data availability. Experiments show that DMNs improve low-resource speech recognition significantly. Moreover, DMNs introduce sparsity to their hidden activations and thus can act as sparse feature extractors.","PeriodicalId":265258,"journal":{"name":"2013 IEEE Workshop on Automatic Speech Recognition and Understanding","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133457477","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2013-12-01DOI: 10.1109/ASRU.2013.6707755
Yosuke Kashiwagi, D. Saito, N. Minematsu, K. Hirose
In this paper, we propose the use of deep neural networks to expand conventional methods of statistical feature enhancement based on piecewise linear transformation. Stereo-based piecewise linear compensation for environments (SPLICE), which is a powerful statistical approach for feature enhancement, models the probabilistic distribution of input noisy features as a mixture of Gaussians. However, soft assignment of an input vector to divided regions is sometimes done inadequately and the vector comes to go through inadequate conversion. Especially when conversion has to be linear, the conversion performance will be easily degraded. Feature enhancement using neural networks is another powerful approach which can directly model a non-linear relationship between noisy and clean feature spaces. In this case, however, it tends to suffer from over-fitting problems. In this paper, we attempt to mitigate this problem by reducing the number of model parameters to estimate. Our neural network is trained whose output layer is associated with the states in the clean feature space, not in the noisy feature space. This strategy makes the size of the output layer independent of the kind of a given noisy environment. Firstly, we characterize the distribution of clean features as a Gaussian mixture model and then, by using deep neural networks, estimate discriminatively the state in the clean space that an input noisy feature corresponds to. Experimental evaluations using the Aurora 2 dataset demonstrate that our proposed method has the best performance compared to conventional methods.
{"title":"Discriminative piecewise linear transformation based on deep learning for noise robust automatic speech recognition","authors":"Yosuke Kashiwagi, D. Saito, N. Minematsu, K. Hirose","doi":"10.1109/ASRU.2013.6707755","DOIUrl":"https://doi.org/10.1109/ASRU.2013.6707755","url":null,"abstract":"In this paper, we propose the use of deep neural networks to expand conventional methods of statistical feature enhancement based on piecewise linear transformation. Stereo-based piecewise linear compensation for environments (SPLICE), which is a powerful statistical approach for feature enhancement, models the probabilistic distribution of input noisy features as a mixture of Gaussians. However, soft assignment of an input vector to divided regions is sometimes done inadequately and the vector comes to go through inadequate conversion. Especially when conversion has to be linear, the conversion performance will be easily degraded. Feature enhancement using neural networks is another powerful approach which can directly model a non-linear relationship between noisy and clean feature spaces. In this case, however, it tends to suffer from over-fitting problems. In this paper, we attempt to mitigate this problem by reducing the number of model parameters to estimate. Our neural network is trained whose output layer is associated with the states in the clean feature space, not in the noisy feature space. This strategy makes the size of the output layer independent of the kind of a given noisy environment. Firstly, we characterize the distribution of clean features as a Gaussian mixture model and then, by using deep neural networks, estimate discriminatively the state in the clean space that an input noisy feature corresponds to. Experimental evaluations using the Aurora 2 dataset demonstrate that our proposed method has the best performance compared to conventional methods.","PeriodicalId":265258,"journal":{"name":"2013 IEEE Workshop on Automatic Speech Recognition and Understanding","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115673587","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2013-12-01DOI: 10.1109/ASRU.2013.6707761
Oliver Walter, Timo Korthals, Reinhold Häb-Umbach, B. Raj
Discovering the linguistic structure of a language solely from spoken input asks for two steps: phonetic and lexical discovery. The first is concerned with identifying the categorical subword unit inventory and relating it to the underlying acoustics, while the second aims at discovering words as repeated patterns of subword units. The hierarchical approach presented here accounts for classification errors in the first stage by modelling the pronunciation of a word in terms of subword units probabilistically: a hidden Markov model with discrete emission probabilities, emitting the observed subword unit sequences. We describe how the system can be learned in a completely unsupervised fashion from spoken input. To improve the initialization of the training of the word pronunciations, the output of a dynamic time warping based acoustic pattern discovery system is used, as it is able to discover similar temporal sequences in the input data. This improved initialization, using only weak supervision, has led to a 40% reduction in word error rate on a digit recognition task.
{"title":"A hierarchical system for word discovery exploiting DTW-based initialization","authors":"Oliver Walter, Timo Korthals, Reinhold Häb-Umbach, B. Raj","doi":"10.1109/ASRU.2013.6707761","DOIUrl":"https://doi.org/10.1109/ASRU.2013.6707761","url":null,"abstract":"Discovering the linguistic structure of a language solely from spoken input asks for two steps: phonetic and lexical discovery. The first is concerned with identifying the categorical subword unit inventory and relating it to the underlying acoustics, while the second aims at discovering words as repeated patterns of subword units. The hierarchical approach presented here accounts for classification errors in the first stage by modelling the pronunciation of a word in terms of subword units probabilistically: a hidden Markov model with discrete emission probabilities, emitting the observed subword unit sequences. We describe how the system can be learned in a completely unsupervised fashion from spoken input. To improve the initialization of the training of the word pronunciations, the output of a dynamic time warping based acoustic pattern discovery system is used, as it is able to discover similar temporal sequences in the input data. This improved initialization, using only weak supervision, has led to a 40% reduction in word error rate on a digit recognition task.","PeriodicalId":265258,"journal":{"name":"2013 IEEE Workshop on Automatic Speech Recognition and Understanding","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130532728","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2013-12-01DOI: 10.1109/ASRU.2013.6707706
Udhyakumar Nallasamy, Mark C. Fuhs, M. Woszczyna, Florian Metze, Tanja Schultz
Speaker dependent (SD) ASR systems have significantly lower word error rates (WER) compared to speaker independent (SI) systems. However, SD systems require sufficient training data from the target speaker, which is impractical to collect in a short time. We present a technique for training SD models using just few minutes of speaker's data. We compensate for the lack of adequate speaker-specific data by selecting neighbours from a database of existing speakers who are acoustically close to the target speaker. These neighbours provide ample training data, which is used to adapt the SI model to obtain an initial SD model for the new speaker with significantly lower WER. We evaluate various neighbour selection algorithms on a large-scale medical transcription task and report significant reduction in WER using only 5 mins of speaker-specific data. We conduct a detailed analysis of various factors such as gender and accent in the neighbour selection. Finally, we study neighbour selection and adaptation in the context of discriminative objective functions.
{"title":"Neighbour selection and adaptation for rapid speaker-dependent ASR","authors":"Udhyakumar Nallasamy, Mark C. Fuhs, M. Woszczyna, Florian Metze, Tanja Schultz","doi":"10.1109/ASRU.2013.6707706","DOIUrl":"https://doi.org/10.1109/ASRU.2013.6707706","url":null,"abstract":"Speaker dependent (SD) ASR systems have significantly lower word error rates (WER) compared to speaker independent (SI) systems. However, SD systems require sufficient training data from the target speaker, which is impractical to collect in a short time. We present a technique for training SD models using just few minutes of speaker's data. We compensate for the lack of adequate speaker-specific data by selecting neighbours from a database of existing speakers who are acoustically close to the target speaker. These neighbours provide ample training data, which is used to adapt the SI model to obtain an initial SD model for the new speaker with significantly lower WER. We evaluate various neighbour selection algorithms on a large-scale medical transcription task and report significant reduction in WER using only 5 mins of speaker-specific data. We conduct a detailed analysis of various factors such as gender and accent in the neighbour selection. Finally, we study neighbour selection and adaptation in the context of discriminative objective functions.","PeriodicalId":265258,"journal":{"name":"2013 IEEE Workshop on Automatic Speech Recognition and Understanding","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129717422","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2013-12-01DOI: 10.1109/ASRU.2013.6707734
A. Ivanov, S. Jalalvand, R. Gretter, D. Falavigna
We explore the impact of speech- and speaker-specific modeling onto the Modulation Spectrum Analysis - Kolmogorov-Smirnov feature Testing (MSA-KST) characterization method in the task of automated prediction of the cognitive impairment diagnosis, namely dysphasia and pervasive development disorder. Phoneme-synchronous capturing of speech dynamics is a reasonable choice for a segmental speech characterization system as it allows comparing speech dynamics in the similar phonetic contexts. Speaker-specific modeling aims at reducing the “within-the-class” variability of the characterized speech or speaker population by removing the effect of speaker properties that should have no relation to the characterization. Specifically the vocal tract length of a speaker has nothing to do with the diagnosis attribution and, thus, the feature set shall be normalized accordingly. The resulting system compares favorably to the baseline system of the Interspeech'2013 Computational Paralinguistics Challenge.
{"title":"Phonetic and anthropometric conditioning of MSA-KST cognitive impairment characterization system","authors":"A. Ivanov, S. Jalalvand, R. Gretter, D. Falavigna","doi":"10.1109/ASRU.2013.6707734","DOIUrl":"https://doi.org/10.1109/ASRU.2013.6707734","url":null,"abstract":"We explore the impact of speech- and speaker-specific modeling onto the Modulation Spectrum Analysis - Kolmogorov-Smirnov feature Testing (MSA-KST) characterization method in the task of automated prediction of the cognitive impairment diagnosis, namely dysphasia and pervasive development disorder. Phoneme-synchronous capturing of speech dynamics is a reasonable choice for a segmental speech characterization system as it allows comparing speech dynamics in the similar phonetic contexts. Speaker-specific modeling aims at reducing the “within-the-class” variability of the characterized speech or speaker population by removing the effect of speaker properties that should have no relation to the characterization. Specifically the vocal tract length of a speaker has nothing to do with the diagnosis attribution and, thus, the feature set shall be normalized accordingly. The resulting system compares favorably to the baseline system of the Interspeech'2013 Computational Paralinguistics Challenge.","PeriodicalId":265258,"journal":{"name":"2013 IEEE Workshop on Automatic Speech Recognition and Understanding","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125468980","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2013-12-01DOI: 10.1109/ASRU.2013.6707732
Duc Le, E. Provost
Research in emotion recognition seeks to develop insights into the temporal properties of emotion. However, automatic emotion recognition from spontaneous speech is challenging due to non-ideal recording conditions and highly ambiguous ground truth labels. Further, emotion recognition systems typically work with noisy high-dimensional data, rendering it difficult to find representative features and train an effective classifier. We tackle this problem by using Deep Belief Networks, which can model complex and non-linear high-level relationships between low-level features. We propose and evaluate a suite of hybrid classifiers based on Hidden Markov Models and Deep Belief Networks. We achieve state-of-the-art results on FAU Aibo, a benchmark dataset in emotion recognition [1]. Our work provides insights into important similarities and differences between speech and emotion.
{"title":"Emotion recognition from spontaneous speech using Hidden Markov models with deep belief networks","authors":"Duc Le, E. Provost","doi":"10.1109/ASRU.2013.6707732","DOIUrl":"https://doi.org/10.1109/ASRU.2013.6707732","url":null,"abstract":"Research in emotion recognition seeks to develop insights into the temporal properties of emotion. However, automatic emotion recognition from spontaneous speech is challenging due to non-ideal recording conditions and highly ambiguous ground truth labels. Further, emotion recognition systems typically work with noisy high-dimensional data, rendering it difficult to find representative features and train an effective classifier. We tackle this problem by using Deep Belief Networks, which can model complex and non-linear high-level relationships between low-level features. We propose and evaluate a suite of hybrid classifiers based on Hidden Markov Models and Deep Belief Networks. We achieve state-of-the-art results on FAU Aibo, a benchmark dataset in emotion recognition [1]. Our work provides insights into important similarities and differences between speech and emotion.","PeriodicalId":265258,"journal":{"name":"2013 IEEE Workshop on Automatic Speech Recognition and Understanding","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126311824","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2013-12-01DOI: 10.1109/ASRU.2013.6707738
A. Touazi, M. Debyeche
This paper proposes a new scheme for low bit-rate source coding of Mel Frequency Cepstral Coefficients (MFCCs) in Distributed Speech Recognition (DSR) system. The method uses the compressed ETSI Advanced Front-End (ETSI-AFE) features factorized into SVD components. By investigating the correlation property between successive MFCC frames, the odd ones are encoded using ETSI-AFE, while only the singular values and the nearest left singular vectors index are encoded and transmitted for the even frames. At the server side, the non-transmitted MFCCs are evaluated through their quantized singular values and the nearest left singular vectors. The system provides a compression bit-rate of 2.7 kbps. The recognition experiments were carried out on the Aurora-2 database for clean and multi-condition training modes. The simulation results show good recognition performance without significant degradation, with respect to the ETSI-AFE encoder.
{"title":"An SVD-based scheme for MFCC compression in distributed speech recognition system","authors":"A. Touazi, M. Debyeche","doi":"10.1109/ASRU.2013.6707738","DOIUrl":"https://doi.org/10.1109/ASRU.2013.6707738","url":null,"abstract":"This paper proposes a new scheme for low bit-rate source coding of Mel Frequency Cepstral Coefficients (MFCCs) in Distributed Speech Recognition (DSR) system. The method uses the compressed ETSI Advanced Front-End (ETSI-AFE) features factorized into SVD components. By investigating the correlation property between successive MFCC frames, the odd ones are encoded using ETSI-AFE, while only the singular values and the nearest left singular vectors index are encoded and transmitted for the even frames. At the server side, the non-transmitted MFCCs are evaluated through their quantized singular values and the nearest left singular vectors. The system provides a compression bit-rate of 2.7 kbps. The recognition experiments were carried out on the Aurora-2 database for clean and multi-condition training modes. The simulation results show good recognition performance without significant degradation, with respect to the ETSI-AFE encoder.","PeriodicalId":265258,"journal":{"name":"2013 IEEE Workshop on Automatic Speech Recognition and Understanding","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126447524","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
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