首页> 外文会议>Mexican International Conference on Artificial Intelligence(MICAI 2007); 20071104-10; Aguascalientes(MX) >Using Adaptive Filter and Wavelets to Increase Automatic Speech Recognition Rate in Noisy Environment
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Using Adaptive Filter and Wavelets to Increase Automatic Speech Recognition Rate in Noisy Environment

机译:在嘈杂环境中使用自适应滤波器和小波提高自动语音识别率

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This paper shows results obtained in the Automatic Speech Recognition (ASR) task for a corpus of digits speech files with a determinate noise level immerse. In the experiments, we used several speech files that contained Gaussian noise. We used HTK (Hidden Markov Model Toolkit) software of Cambridge University in the experiments. The noise level added to the speech signals was varying from fifteen to forty Db increased by a step of 5 units. We used an adaptive filtering to reduce the level noise (it was based in the Least Measure Square -LMS- algorithm) and two different wavelets (Haar and Daubechies). With LMS we obtained an error rate lower than if it was not present and it was better than wavelets employed for this experiment of Automatic Speech Recognition. For decreasing the error rate we trained with 50% of contaminated and originals signals to the ASR system. The results showed in this paper are focused to try analyses the ASR performance in a noisy environment and to demonstrate that if we are controlling the noise level and if we know the application where it is going to work, then we can obtain a better response in the ASR tasks. Is very interesting to count with these results because speech signal that we can find in a real experiment (extracted from an environment work, I.e.), could be treated with these technique and we can decrease the error rate obtained. Finally, we report a recognition rate of 99%, 97.5% 96%, 90.5%, 81% and 78.5% obtained from 15, 20, 25, 30, 35 and 40 noise levels, respectively when the corpus mentioned before was employed and LMS algorithm was used. Haar wavelet level 1 reached up the most important results as an alternative to LMS algorithm, but only when the noise level was 40 Db and using original corpus.
机译:本文显示了在自动语音识别(ASR)任务中获得的具有确定噪声水平的数字语音文件集的结果。在实验中,我们使用了几个包含高斯噪声的语音文件。在实验中,我们使用了剑桥大学的HTK(隐马尔可夫模型工具包)软件。添加到语音信号中的噪声水平从15到40 Db不等,增加了5个单位。我们使用了自适应滤波来降低电平噪声(它是基于最小测量平方-LMS-算法的)和两个不同的小波(Haar和Daubechies)。使用LMS,我们获得的错误率低于不存在的错误率,并且比用于自动语音识别实验的小波要好。为了降低错误率,我们使用了50%的污染信号和原始信号训练了ASR系统。本文显示的结果着重于尝试分析嘈杂环境中的ASR性能,并证明如果我们正在控制噪声水平,并且如果我们知道应用程序将在哪里工作,那么我们可以获得更好的响应。 ASR任务。用这些结果来计数是非常有趣的,因为可以使用这些技术来处理我们在真实实验中发现的语音信号(从环境工作中提取),并且可以降低获得的错误率。最后,我们报告当使用上述语料库和LMS时,分别从15、20、25、30、35和40噪声水平获得的识别率分别为99%,97.5%,96%,90.5%,81%和78.5%。使用算法。 Haar小波1级达到了最重要的结果,可以替代LMS算法,但仅当噪声级为40 Db且使用原始语料库时才达到。

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