3,178 research outputs found
Feature Extracting in the Presence of Environmental Noise, using Subband Adaptive Filtering
In this work, a new feature extracting method in noisy environments is proposed. The approach is based on subband decomposition of speech signals followed by adaptive filtering in the noisiest subbbands of speech. The speech decomposition is obtained using low complexity octave filter bank, while adaptive filtering is performed using the normalized least mean square algorithm. The performance of the new feature was evaluated for isolated word speech recognition in the presence of a car noise. The proposed method showed higher recognition accuracy than conventional methods in noisy environments
The HTS-2008 System: Yet Another Evaluation of the Speaker-Adaptive HMM-based Speech Synthesis System in The 2008 Blizzard Challenge
For the 2008 Blizzard Challenge, we used the same speaker-adaptive approach to HMM-based speech synthesis that was used in the HTS entry to the 2007 challenge, but an improved system was built in which the multi-accented English average voice model was trained on 41 hours of speech data with high-order mel-cepstral analysis using an efficient forward-backward algorithm for the HSMM. The listener evaluation scores for the synthetic speech generated from this system was much better than in 2007: the system had the equal best naturalness on the small English data set and the equal best intelligibility on both small and large data sets for English, and had the equal best naturalness on the Mandarin data. In fact, the English system was found to be as intelligible as human speech
Voice Conversion Using Sequence-to-Sequence Learning of Context Posterior Probabilities
Voice conversion (VC) using sequence-to-sequence learning of context
posterior probabilities is proposed. Conventional VC using shared context
posterior probabilities predicts target speech parameters from the context
posterior probabilities estimated from the source speech parameters. Although
conventional VC can be built from non-parallel data, it is difficult to convert
speaker individuality such as phonetic property and speaking rate contained in
the posterior probabilities because the source posterior probabilities are
directly used for predicting target speech parameters. In this work, we assume
that the training data partly include parallel speech data and propose
sequence-to-sequence learning between the source and target posterior
probabilities. The conversion models perform non-linear and variable-length
transformation from the source probability sequence to the target one. Further,
we propose a joint training algorithm for the modules. In contrast to
conventional VC, which separately trains the speech recognition that estimates
posterior probabilities and the speech synthesis that predicts target speech
parameters, our proposed method jointly trains these modules along with the
proposed probability conversion modules. Experimental results demonstrate that
our approach outperforms the conventional VC.Comment: Accepted to INTERSPEECH 201
Analysis of Speaker Clustering Strategies for HMM-Based Speech Synthesis
This paper describes a method for speaker clustering, with the application of building average voice models for speakeradaptive HMM-based speech synthesis that are a good basis for adapting to specific target speakers. Our main hypothesis is that using perceptually similar speakers to build the average voice model will be better than use unselected speakers, even if the amount of data available from perceptually similar speakers is smaller. We measure the perceived similarities among a group of 30 female speakers in a listening test and then apply multiple linear regression to automatically predict these listener judgements of speaker similarity and thus to identify similar speakers automatically. We then compare a variety of average voice models trained on either speakers who were perceptually judged to be similar to the target speaker, or speakers selected by the multiple linear regression, or a large global set of unselected speakers. We find that the average voice model trained on perceptually similar speakers provides better performance than the global model, even though the latter is trained on more data, confirming our main hypothesis. However, the average voice model using speakers selected automatically by the multiple linear regression does not reach the same level of performance. Index Terms: Statistical parametric speech synthesis, hidden Markov models, speaker adaptatio
Anti-spoofing Methods for Automatic SpeakerVerification System
Growing interest in automatic speaker verification (ASV)systems has lead to
significant quality improvement of spoofing attackson them. Many research works
confirm that despite the low equal er-ror rate (EER) ASV systems are still
vulnerable to spoofing attacks. Inthis work we overview different acoustic
feature spaces and classifiersto determine reliable and robust countermeasures
against spoofing at-tacks. We compared several spoofing detection systems,
presented so far,on the development and evaluation datasets of the Automatic
SpeakerVerification Spoofing and Countermeasures (ASVspoof) Challenge
2015.Experimental results presented in this paper demonstrate that the useof
magnitude and phase information combination provides a substantialinput into
the efficiency of the spoofing detection systems. Also wavelet-based features
show impressive results in terms of equal error rate. Inour overview we compare
spoofing performance for systems based on dif-ferent classifiers. Comparison
results demonstrate that the linear SVMclassifier outperforms the conventional
GMM approach. However, manyresearchers inspired by the great success of deep
neural networks (DNN)approaches in the automatic speech recognition, applied
DNN in thespoofing detection task and obtained quite low EER for known and
un-known type of spoofing attacks.Comment: 12 pages, 0 figures, published in Springer Communications in Computer
and Information Science (CCIS) vol. 66
- …