Pub Date : 2012-12-01DOI: 10.1109/SLT.2012.6424219
Hung-yi Lee, Tsung-Hsien Wen, Lin-Shan Lee
Retrieving objects semantically related to the query has been widely studied in text information retrieval. However, when applying the text-based techniques on spoken content, the inevitable recognition errors may seriously degrade the performance. In this paper, we propose to enhance the expected term frequencies estimated from spoken content by acoustic similarity graphs. For each word in the lexicon, a graph is constructed describing acoustic similarity among spoken segments in the archive. Score propagation over the graph helps in estimating the expected term frequencies. The enhanced expected term frequencies can be used in the language modeling retrieval approach, as well as semantic retrieval techniques such as the document expansion based on latent semantic analysis, and query expansion considering both words and latent topic information. Preliminary experiments performed on Mandarin broadcast news indicated that improved performance were achievable under different conditions.
{"title":"Improved semantic retrieval of spoken content by language models enhanced with acoustic similarity graph","authors":"Hung-yi Lee, Tsung-Hsien Wen, Lin-Shan Lee","doi":"10.1109/SLT.2012.6424219","DOIUrl":"https://doi.org/10.1109/SLT.2012.6424219","url":null,"abstract":"Retrieving objects semantically related to the query has been widely studied in text information retrieval. However, when applying the text-based techniques on spoken content, the inevitable recognition errors may seriously degrade the performance. In this paper, we propose to enhance the expected term frequencies estimated from spoken content by acoustic similarity graphs. For each word in the lexicon, a graph is constructed describing acoustic similarity among spoken segments in the archive. Score propagation over the graph helps in estimating the expected term frequencies. The enhanced expected term frequencies can be used in the language modeling retrieval approach, as well as semantic retrieval techniques such as the document expansion based on latent semantic analysis, and query expansion considering both words and latent topic information. Preliminary experiments performed on Mandarin broadcast news indicated that improved performance were achievable under different conditions.","PeriodicalId":375378,"journal":{"name":"2012 IEEE Spoken Language Technology Workshop (SLT)","volume":"72 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2012-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122277832","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 : 2012-12-01DOI: 10.1109/SLT.2012.6424214
G. Anumanchipalli, Luís C. Oliveira, A. Black
This paper presents an approach for transfer of speaker intent in speech-to-speech machine translation (S2SMT). Specifically, we describe techniques to retain the prominence patterns of the source language utterance through the translation pipeline and impose this information during speech synthesis in the target language. We first present an analysis of word focus across languages to motivate the problem of transfer. We then propose an approach for training an appropriate transfer function for intonation on a parallel speech corpus in the two languages within which the translation is carried out. We present our analysis and experiments on English↔Portuguese and English↔German language pairs and evaluate the proposed transformation techniques through objective measures.
{"title":"Intent transfer in speech-to-speech machine translation","authors":"G. Anumanchipalli, Luís C. Oliveira, A. Black","doi":"10.1109/SLT.2012.6424214","DOIUrl":"https://doi.org/10.1109/SLT.2012.6424214","url":null,"abstract":"This paper presents an approach for transfer of speaker intent in speech-to-speech machine translation (S2SMT). Specifically, we describe techniques to retain the prominence patterns of the source language utterance through the translation pipeline and impose this information during speech synthesis in the target language. We first present an analysis of word focus across languages to motivate the problem of transfer. We then propose an approach for training an appropriate transfer function for intonation on a parallel speech corpus in the two languages within which the translation is carried out. We present our analysis and experiments on English↔Portuguese and English↔German language pairs and evaluate the proposed transformation techniques through objective measures.","PeriodicalId":375378,"journal":{"name":"2012 IEEE Spoken Language Technology Workshop (SLT)","volume":"60 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2012-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124691012","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 : 2012-12-01DOI: 10.1109/SLT.2012.6424220
Tsung-Hsien Wen, Hung-yi Lee, Tai-Yuan Chen, Lin-Shan Lee
Voice access of cloud applications via smartphones is very attractive today, specifically because a smartphones is used by a single user, so personalized acoustic/language models become feasible. In particular, huge quantities of texts are available within the social networks over the Internet with known authors and given relationships, it is possible to train personalized language models because it is reasonable to assume users with those relationships may share some common subject topics, wording habits and linguistic patterns. In this paper, we propose an adaptation framework for building a robust personalized language model by incorporating the texts the target user and other users had posted on the social networks over the Internet to take care of the linguistic mismatch across different users. Experiments on Facebook dataset showed encouraging improvements in terms of both model perplexity and recognition accuracy with proposed approaches considering relationships among users, similarity based on latent topics, and random walk over a user graph.
{"title":"Personalized language modeling by crowd sourcing with social network data for voice access of cloud applications","authors":"Tsung-Hsien Wen, Hung-yi Lee, Tai-Yuan Chen, Lin-Shan Lee","doi":"10.1109/SLT.2012.6424220","DOIUrl":"https://doi.org/10.1109/SLT.2012.6424220","url":null,"abstract":"Voice access of cloud applications via smartphones is very attractive today, specifically because a smartphones is used by a single user, so personalized acoustic/language models become feasible. In particular, huge quantities of texts are available within the social networks over the Internet with known authors and given relationships, it is possible to train personalized language models because it is reasonable to assume users with those relationships may share some common subject topics, wording habits and linguistic patterns. In this paper, we propose an adaptation framework for building a robust personalized language model by incorporating the texts the target user and other users had posted on the social networks over the Internet to take care of the linguistic mismatch across different users. Experiments on Facebook dataset showed encouraging improvements in terms of both model perplexity and recognition accuracy with proposed approaches considering relationships among users, similarity based on latent topics, and random walk over a user graph.","PeriodicalId":375378,"journal":{"name":"2012 IEEE Spoken Language Technology Workshop (SLT)","volume":"53 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2012-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130066166","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 : 2012-12-01DOI: 10.1109/SLT.2012.6424204
Nicolas Obin, M. Liuni
This paper introduces an entropy-based spectral representation as a measure of the degree of noisiness in audio signals, complementary to the standard MFCCs for audio and speech recognition. The proposed representation is based on the Rényi entropy, which is a generalization of the Shannon entropy. In audio signal representation, Rényi entropy presents the advantage of focusing either on the harmonic content (prominent amplitude within a distribution) or on the noise content (equal distribution of amplitudes). The proposed representation outperforms all other noisiness measures - including Shannon and Wiener entropies - in a large-scale classification of vocal effort (whispered-soft/normal/loud-shouted) in the real scenario of multi-language massive role-playing video games. The improvement is around 10% in relative error reduction, and is particularly significant for the recognition of noisy speech - i.e., whispery/breathy speech. This confirms the role of noisiness for speech recognition, and will further be extended to the classification of voice quality for the design of an automatic voice casting system in video games.
{"title":"On the generalization of Shannon entropy for speech recognition","authors":"Nicolas Obin, M. Liuni","doi":"10.1109/SLT.2012.6424204","DOIUrl":"https://doi.org/10.1109/SLT.2012.6424204","url":null,"abstract":"This paper introduces an entropy-based spectral representation as a measure of the degree of noisiness in audio signals, complementary to the standard MFCCs for audio and speech recognition. The proposed representation is based on the Rényi entropy, which is a generalization of the Shannon entropy. In audio signal representation, Rényi entropy presents the advantage of focusing either on the harmonic content (prominent amplitude within a distribution) or on the noise content (equal distribution of amplitudes). The proposed representation outperforms all other noisiness measures - including Shannon and Wiener entropies - in a large-scale classification of vocal effort (whispered-soft/normal/loud-shouted) in the real scenario of multi-language massive role-playing video games. The improvement is around 10% in relative error reduction, and is particularly significant for the recognition of noisy speech - i.e., whispery/breathy speech. This confirms the role of noisiness for speech recognition, and will further be extended to the classification of voice quality for the design of an automatic voice casting system in video games.","PeriodicalId":375378,"journal":{"name":"2012 IEEE Spoken Language Technology Workshop (SLT)","volume":"159 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2012-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126208023","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 : 2012-12-01DOI: 10.1109/SLT.2012.6424258
Heather Friedberg, D. Litman, Susannah B. F. Paletz
Lexical entrainment is a measure of how the words that speakers use in a conversation become more similar over time. In this paper, we propose a measure of lexical entrainment for multi-party speaking situations. We apply this score to a corpus of student engineering groups using high-frequency words and project words, and investigate the relationship between lexical entrainment and group success on a class project. Our initial findings show that, using the entrainment score with project-related words, there is a significant difference between the lexical entrainment of high performing groups, which tended to increase with time, and the entrainment for low performing groups, which tended to decrease with time.
{"title":"Lexical entrainment and success in student engineering groups","authors":"Heather Friedberg, D. Litman, Susannah B. F. Paletz","doi":"10.1109/SLT.2012.6424258","DOIUrl":"https://doi.org/10.1109/SLT.2012.6424258","url":null,"abstract":"Lexical entrainment is a measure of how the words that speakers use in a conversation become more similar over time. In this paper, we propose a measure of lexical entrainment for multi-party speaking situations. We apply this score to a corpus of student engineering groups using high-frequency words and project words, and investigate the relationship between lexical entrainment and group success on a class project. Our initial findings show that, using the entrainment score with project-related words, there is a significant difference between the lexical entrainment of high performing groups, which tended to increase with time, and the entrainment for low performing groups, which tended to decrease with time.","PeriodicalId":375378,"journal":{"name":"2012 IEEE Spoken Language Technology Workshop (SLT)","volume":"36 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2012-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125009370","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 : 2012-12-01DOI: 10.1109/SLT.2012.6424260
Sandrine Brognaux, Sophie Roekhaut, Thomas Drugman, Richard Beaufort
Several automatic phonetic alignment tools have been proposed in the literature. They usually rely on pre-trained speaker-independent models to align new corpora. Their drawback is that they cover a very limited number of languages and might not perform properly for different speaking styles. This paper presents a new tool for automatic phonetic alignment available online. Its specificity is that it trains the model directly on the corpus to align, which makes it applicable to any language and speaking style. Experiments on three corpora show that it provides results comparable to other existing tools. It also allows the tuning of some training parameters. The use of tied-state triphones, for example, shows further improvement of about 1.5% for a 20 ms threshold. A manually-aligned part of the corpus can also be used as bootstrap to improve the model quality. Alignment rates were found to significantly increase, up to 20%, using only 30 seconds of bootstrapping data.
{"title":"Train&align: A new online tool for automatic phonetic alignment","authors":"Sandrine Brognaux, Sophie Roekhaut, Thomas Drugman, Richard Beaufort","doi":"10.1109/SLT.2012.6424260","DOIUrl":"https://doi.org/10.1109/SLT.2012.6424260","url":null,"abstract":"Several automatic phonetic alignment tools have been proposed in the literature. They usually rely on pre-trained speaker-independent models to align new corpora. Their drawback is that they cover a very limited number of languages and might not perform properly for different speaking styles. This paper presents a new tool for automatic phonetic alignment available online. Its specificity is that it trains the model directly on the corpus to align, which makes it applicable to any language and speaking style. Experiments on three corpora show that it provides results comparable to other existing tools. It also allows the tuning of some training parameters. The use of tied-state triphones, for example, shows further improvement of about 1.5% for a 20 ms threshold. A manually-aligned part of the corpus can also be used as bootstrap to improve the model quality. Alignment rates were found to significantly increase, up to 20%, using only 30 seconds of bootstrapping data.","PeriodicalId":375378,"journal":{"name":"2012 IEEE Spoken Language Technology Workshop (SLT)","volume":"142 6 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2012-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114217865","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 : 2012-12-01DOI: 10.1109/SLT.2012.6424259
Sandrine Brognaux, Thomas Drugman, Richard Beaufort
Both unit-selection and HMM-based speech synthesis require large annotated speech corpora. To generate more natural speech, considering the prosodic nature of each phoneme of the corpus is crucial. Generally, phonemes are assigned labels which should reflect their suprasegmental characteristics. Labels often result from an automatic syntactic analysis, without checking the acoustic realization of the phoneme in the corpus. This leads to numerous errors because syntax and prosody do not always coincide. This paper proposes a method to reduce the amount of labeling errors, using acoustic information. It is applicable as a post-process to any syntax-driven prosody labeling. Acoustic features are considered, to check the syntax-based labels and suggest potential modifications. The proposed technique has the advantage of not requiring a manually prosody-labelled corpus. The evaluation on a corpus in French shows that more than 75% of the errors detected by the method are effective errors which must be corrected.
{"title":"Automatic detection and correction of syntax-based prosody annotation errors","authors":"Sandrine Brognaux, Thomas Drugman, Richard Beaufort","doi":"10.1109/SLT.2012.6424259","DOIUrl":"https://doi.org/10.1109/SLT.2012.6424259","url":null,"abstract":"Both unit-selection and HMM-based speech synthesis require large annotated speech corpora. To generate more natural speech, considering the prosodic nature of each phoneme of the corpus is crucial. Generally, phonemes are assigned labels which should reflect their suprasegmental characteristics. Labels often result from an automatic syntactic analysis, without checking the acoustic realization of the phoneme in the corpus. This leads to numerous errors because syntax and prosody do not always coincide. This paper proposes a method to reduce the amount of labeling errors, using acoustic information. It is applicable as a post-process to any syntax-driven prosody labeling. Acoustic features are considered, to check the syntax-based labels and suggest potential modifications. The proposed technique has the advantage of not requiring a manually prosody-labelled corpus. The evaluation on a corpus in French shows that more than 75% of the errors detected by the method are effective errors which must be corrected.","PeriodicalId":375378,"journal":{"name":"2012 IEEE Spoken Language Technology Workshop (SLT)","volume":"119 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2012-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114330805","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 : 2012-12-01DOI: 10.1109/SLT.2012.6424216
Md. Akmal Haidar, D. O'Shaughnessy
We introduce novel language model (LM) adaptation approaches using the latent Dirichlet allocation (LDA) model. Observed n-grams in the training set are assigned to topics using soft and hard clustering. In soft clustering, each n-gram is assigned to topics such that the total count of that n-gram for all topics is equal to the global count of that n-gram in the training set. Here, the normalized topic weights of the n-gram are multiplied by the global n-gram count to form the topic n-gram count for the respective topics. In hard clustering, each n-gram is assigned to a single topic with the maximum fraction of the global n-gram count for the corresponding topic. Here, the topic is selected using the maximum topic weight for the n-gram. The topic n-gram count LMs are created using the respective topic n-gram counts and adapted by using the topic weights of a development test set. We compute the average of the confidence measures: the probability of word given topic and the probability of topic given word. The average is taken over the words in the n-grams and the development test set to form the topic weights of the n-grams and the development test set respectively. Our approaches show better performance over some traditional approaches using the WSJ corpus.
{"title":"Topic n-gram count language model adaptation for speech recognition","authors":"Md. Akmal Haidar, D. O'Shaughnessy","doi":"10.1109/SLT.2012.6424216","DOIUrl":"https://doi.org/10.1109/SLT.2012.6424216","url":null,"abstract":"We introduce novel language model (LM) adaptation approaches using the latent Dirichlet allocation (LDA) model. Observed n-grams in the training set are assigned to topics using soft and hard clustering. In soft clustering, each n-gram is assigned to topics such that the total count of that n-gram for all topics is equal to the global count of that n-gram in the training set. Here, the normalized topic weights of the n-gram are multiplied by the global n-gram count to form the topic n-gram count for the respective topics. In hard clustering, each n-gram is assigned to a single topic with the maximum fraction of the global n-gram count for the corresponding topic. Here, the topic is selected using the maximum topic weight for the n-gram. The topic n-gram count LMs are created using the respective topic n-gram counts and adapted by using the topic weights of a development test set. We compute the average of the confidence measures: the probability of word given topic and the probability of topic given word. The average is taken over the words in the n-grams and the development test set to form the topic weights of the n-grams and the development test set respectively. Our approaches show better performance over some traditional approaches using the WSJ corpus.","PeriodicalId":375378,"journal":{"name":"2012 IEEE Spoken Language Technology Workshop (SLT)","volume":"4 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2012-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130771343","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 : 2012-12-01DOI: 10.1109/SLT.2012.6424205
Shuji Komeiji, T. Arakawa, Takafumi Koshinaka
This paper proposes a noise-robust speech recognition method composed of weak noise suppression (NS) and weak Vector Taylor Series Adaptation (VTSA). The proposed method compensates defects of NS and VTSA, and gains only the advantages by them. The weak NS reduces distortion by over-suppression that may accompany noise-suppressed speech. The weak VTSA avoids over-adaptation by offsetting a part of acoustic-model adaptation that corresponds to the suppressed noise. Evaluation results with the AURORA2 database show that the proposed method achieves as much as 1.2 points higher word accuracy (87.4%) than a method with VTSA alone (86.2%) that is always better than its counterpart with NS.
{"title":"A noise-robust speech recognition method composed of weak noise suppression and weak Vector Taylor Series Adaptation","authors":"Shuji Komeiji, T. Arakawa, Takafumi Koshinaka","doi":"10.1109/SLT.2012.6424205","DOIUrl":"https://doi.org/10.1109/SLT.2012.6424205","url":null,"abstract":"This paper proposes a noise-robust speech recognition method composed of weak noise suppression (NS) and weak Vector Taylor Series Adaptation (VTSA). The proposed method compensates defects of NS and VTSA, and gains only the advantages by them. The weak NS reduces distortion by over-suppression that may accompany noise-suppressed speech. The weak VTSA avoids over-adaptation by offsetting a part of acoustic-model adaptation that corresponds to the suppressed noise. Evaluation results with the AURORA2 database show that the proposed method achieves as much as 1.2 points higher word accuracy (87.4%) than a method with VTSA alone (86.2%) that is always better than its counterpart with NS.","PeriodicalId":375378,"journal":{"name":"2012 IEEE Spoken Language Technology Workshop (SLT)","volume":"19 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2012-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128884900","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 : 2012-12-01DOI: 10.1109/SLT.2012.6424211
Cong-Thanh Do, M. Taghizadeh, Philip N. Garner
This paper investigates the combination of cepstral normalization and cochlear implant-like speech processing for microphone array-based speech recognition. Testing speech signals are recorded by a circular microphone array and are subsequently processed with superdirective beamforming and McCowan post-filtering. Training speech signals, from the multichannel overlapping Number corpus (MONC), are clean and not overlapping. Cochlear implant-like speech processing, which is inspired from the speech processing strategy in cochlear implants, is applied on the training and testing speech signals. Cepstral normalization, including cepstral mean and variance normalization (CMN and CVN), are applied on the training and testing cepstra. Experiments show that implementing either cepstral normalization or cochlear implant-like speech processing helps in reducing the WERs of microphone array-based speech recognition. Combining cepstral normalization and cochlear implant-like speech processing reduces further the WERs, when there is overlapping speech. Train/test mismatches are measured using the Kullback-Leibler divergence (KLD), between the global probability density functions (PDFs) of training and testing cepstral vectors. This measure reveals a train/test mismatch reduction when either cepstral normalization or cochlear implant-like speech processing is used. It reveals also that combining these two processing reduces further the train/test mismatches as well as the WERs.
{"title":"Combining cepstral normalization and cochlear implant-like speech processing for microphone array-based speech recognition","authors":"Cong-Thanh Do, M. Taghizadeh, Philip N. Garner","doi":"10.1109/SLT.2012.6424211","DOIUrl":"https://doi.org/10.1109/SLT.2012.6424211","url":null,"abstract":"This paper investigates the combination of cepstral normalization and cochlear implant-like speech processing for microphone array-based speech recognition. Testing speech signals are recorded by a circular microphone array and are subsequently processed with superdirective beamforming and McCowan post-filtering. Training speech signals, from the multichannel overlapping Number corpus (MONC), are clean and not overlapping. Cochlear implant-like speech processing, which is inspired from the speech processing strategy in cochlear implants, is applied on the training and testing speech signals. Cepstral normalization, including cepstral mean and variance normalization (CMN and CVN), are applied on the training and testing cepstra. Experiments show that implementing either cepstral normalization or cochlear implant-like speech processing helps in reducing the WERs of microphone array-based speech recognition. Combining cepstral normalization and cochlear implant-like speech processing reduces further the WERs, when there is overlapping speech. Train/test mismatches are measured using the Kullback-Leibler divergence (KLD), between the global probability density functions (PDFs) of training and testing cepstral vectors. This measure reveals a train/test mismatch reduction when either cepstral normalization or cochlear implant-like speech processing is used. It reveals also that combining these two processing reduces further the train/test mismatches as well as the WERs.","PeriodicalId":375378,"journal":{"name":"2012 IEEE Spoken Language Technology Workshop (SLT)","volume":"134 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2012-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116829370","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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