Filterbank learning using Convolutional Restricted Boltzmann Machine for speech recognition

Hardik B. Sailor, H. Patil
{"title":"Filterbank learning using Convolutional Restricted Boltzmann Machine for speech recognition","authors":"Hardik B. Sailor, H. Patil","doi":"10.1109/ICASSP.2016.7472808","DOIUrl":null,"url":null,"abstract":"Convolutional Restricted Boltzmann Machine (ConvRBM) as a model for speech signal is presented in this paper. We have developed ConvRBM with sampling from noisy rectified linear units (NReLUs). ConvRBM is trained in an unsupervised way to model speech signal of arbitrary lengths. Weights of the model can represent an auditory-like filterbank. Our proposed learned filterbank is also nonlinear with respect to center frequencies of subband filters similar to standard filterbanks (such as Mel, Bark, ERB, etc.). We have used our proposed model as a front-end to learn features and applied to speech recognition task. Performance of ConvRBM features is improved compared to MFCC with relative improvement of 5% on TIMIT test set and 7% on WSJ0 database for both Nov'92 test sets using GMM-HMM systems. With DNN-HMM systems, we achieved relative improvement of 3% on TIMIT test set over MFCC and Mel filterbank (FBANK). On WSJ0 Nov'92 test sets, we achieved relative improvement of 4-14% using ConvRBM features over MFCC features and 3.6-5.6% using ConvRBM filterbank over FBANK features.","PeriodicalId":165321,"journal":{"name":"2016 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","volume":"10 1","pages":"0"},"PeriodicalIF":0.0000,"publicationDate":"2016-03-31","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":"28","resultStr":null,"platform":"Semanticscholar","paperid":null,"PeriodicalName":"2016 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)","FirstCategoryId":"1085","ListUrlMain":"https://doi.org/10.1109/ICASSP.2016.7472808","RegionNum":0,"RegionCategory":null,"ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":null,"EPubDate":"","PubModel":"","JCR":"","JCRName":"","Score":null,"Total":0}
引用次数: 28

Abstract

Convolutional Restricted Boltzmann Machine (ConvRBM) as a model for speech signal is presented in this paper. We have developed ConvRBM with sampling from noisy rectified linear units (NReLUs). ConvRBM is trained in an unsupervised way to model speech signal of arbitrary lengths. Weights of the model can represent an auditory-like filterbank. Our proposed learned filterbank is also nonlinear with respect to center frequencies of subband filters similar to standard filterbanks (such as Mel, Bark, ERB, etc.). We have used our proposed model as a front-end to learn features and applied to speech recognition task. Performance of ConvRBM features is improved compared to MFCC with relative improvement of 5% on TIMIT test set and 7% on WSJ0 database for both Nov'92 test sets using GMM-HMM systems. With DNN-HMM systems, we achieved relative improvement of 3% on TIMIT test set over MFCC and Mel filterbank (FBANK). On WSJ0 Nov'92 test sets, we achieved relative improvement of 4-14% using ConvRBM features over MFCC features and 3.6-5.6% using ConvRBM filterbank over FBANK features.
查看原文
分享 分享
微信好友 朋友圈 QQ好友 复制链接
本刊更多论文
使用卷积受限玻尔兹曼机进行语音识别的滤波器组学习
本文提出了卷积受限玻尔兹曼机(ConvRBM)作为语音信号的模型。我们开发了从噪声整流线性单元(NReLUs)采样的卷积rbm。采用无监督的方法训练卷积rbm,对任意长度的语音信号进行建模。模型的权重可以表示一个类似听觉的滤波器组。我们提出的学习滤波器组对于子带滤波器的中心频率也是非线性的,类似于标准滤波器组(如Mel, Bark, ERB等)。我们将所提出的模型作为学习特征的前端,并应用于语音识别任务。与MFCC相比,使用GMM-HMM系统的ConvRBM特征的性能得到了提高,在TIMIT测试集上的相对提高了5%,在WSJ0数据库上的相对提高了7%。对于DNN-HMM系统,我们在TIMIT测试集上比MFCC和Mel滤波器组(FBANK)实现了3%的相对改进。在WSJ0 11月92日的测试集上,我们使用ConvRBM特征比MFCC特征获得了4-14%的相对改进,使用ConvRBM滤波器组比FBANK特征获得了3.6-5.6%的相对改进。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
求助全文
约1分钟内获得全文 去求助
来源期刊
自引率
0.00%
发文量
0
期刊最新文献
Self-stabilized deep neural network An acoustic keystroke transient canceler for speech communication terminals using a semi-blind adaptive filter model Data sketching for large-scale Kalman filtering Improved decoding of analog modulo block codes for noise mitigation An expectation-maximization eigenvector clustering approach to direction of arrival estimation of multiple speech sources
×
引用
GB/T 7714-2015
复制
MLA
复制
APA
复制
导出至
BibTeX EndNote RefMan NoteFirst NoteExpress
×
×
提示
您的信息不完整,为了账户安全,请先补充。
现在去补充
×
提示
您因"违规操作"
具体请查看互助需知
我知道了
×
提示
现在去查看 取消
×
提示
确定
0
微信
客服QQ
Book学术公众号 扫码关注我们
反馈
×
意见反馈
请填写您的意见或建议
请填写您的手机或邮箱
已复制链接
已复制链接
快去分享给好友吧!
我知道了
×
扫码分享
扫码分享
Book学术官方微信
Book学术文献互助
Book学术文献互助群
群 号:481959085
Book学术
文献互助 智能选刊 最新文献 互助须知 联系我们:info@booksci.cn
Book学术提供免费学术资源搜索服务,方便国内外学者检索中英文文献。致力于提供最便捷和优质的服务体验。
Copyright © 2023 Book学术 All rights reserved.
ghs 京公网安备 11010802042870号 京ICP备2023020795号-1