2015 · 13 citations · 13 references
ComputingSpeech-language PathologyAcousticsComputational AcousticsMachine LearningDeep LearningSpeech AcousticsSpeech Intelligibility EnhancementSpeech RecognitionAutomatic RecognitionDeep Neural Network Acoustic ModelsMulti-microphone Signal ProcessingRobust Speech RecognitionSpeaker AdaptationSpeech Recognition AccuracySpeech Recognition EvaluationWord Error Rate
In this paper, we introduce the system developed at the Institute for Infocomm Research (I2 R) for the ASpIRE (Automatic Speech recognition In Reverberant Environments) challenge. The main components of the system are a front-end processing system consisting of a distributed beam-forming algorithm, that performs adaptive weighting and channel elimination, a speech dereverberation approach using a maximum-kurtosis criteria, and a robust voice activity detection (VAD) module based on using the sub-harmonic ratio (SHR). The acoustic back-end consists of a multi-conditional Deep Neural Network (DNN) model that uses speaker adapted features combined with a decoding strategy that performs semi-supervised DNN model adaptation using weighted labels generated by the first-pass decoding output. On the single-microphone evaluation, our system achieved a word error rate (WER) of 44.8%. With the incorporation of beamforming on the multi-microphone evaluation, our system achieved an improvement in WER of over 6% to give the best evaluation result of 38.5%.
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Kaldi Speech Recognition Toolkit
Daniel Povey · Infoscience (Ecole Polytechnique Fédérale de Lausanne) · 2024 · 4.9K citations · Full text
Computing, Linguistics, Acoustics +15
Sequence-discriminative training of deep neural networks
Karel Veselý, Arnab Ghoshal, Lukáš Burget et al. · 2013 · 662 citations · Full text