1,416 research outputs found
Making Reassignment Adjustable: the Levenberg-Marquardt Approach
accepted for publication, to appear in Proc. of IEEE Int. Conf. on Acoust., Speech and Signal Proc. ICASSP-12, Kyoto (Japan), March 25-30, 2012.International audienceThis paper presents a new time-frequency reassignment process of the spectrogram, called the Levenberg-Marquardt reassignment. Compared to the classical one, this new reassignment process uses the second-order derivatives of the phase of the short-time Fourier transform, and provides the user with a setting parameter. This parameter allows him to produce either a weaker or a stronger localization of the signal components in the time-frequency plane
Cepstral analysis based on the Glimpse proportion measure for improving the intelligibility of HMM-based synthetic speech in noise
In this paper we introduce a new cepstral coefficient extraction method based on an intelligibility measure for speech in noise, the Glimpse Proportion measure. This new method aims to increase the intelligibility of speech in noise by modifying the clean speech, and has applications in scenarios such as public announcement and car navigation systems. We first explain how the Glimpse Proportion measure operates and further show how we approximated it to integrate it into an existing spectral envelope parameter extraction method commonly used in the HMM-based speech synthesis framework. We then demonstrate how this new method changes the modelled spectrum according to the characteristics of the noise and show results for a listening test with vocoded and HMM-based synthetic speech. The test indicates that the proposed method can significantly improve intelligibility of synthetic speech in speech shaped noise. Index Terms — cepstral coefficient extraction, objective measure for speech intelligibility, Lombard speech, HMM-based speech synthesis 1
On the effect of SNR and superdirective beamforming in speaker diarisation in meetings
This paper examines the effect of sensor performance on speaker diarisation in meetings and investigates the use of more advanced beamforming techniques, beyond the typically employed delay-sum beamformer, for mitigating the effects of poorer sensor performance. We present superdirective beamforming and investigate how different time difference of arrival (TDOA) smoothing and beamforming techniques influence the performance of state-of-the-art diarisation systems. We produced and transcribed a new corpus of meetings recorded in the instrumented meeting room using a high SNR analogue and a newly developed low SNR digital MEMS microphone array (DMMA.2). This research demonstrates that TDOA smoothing has a significant effect on the diarisation error rate and that simple noise reduction and beamforming schemes suffice to overcome audio signal degradation due to the lower SNR of modern MEMS microphones. Index Terms — Speaker diarisation in meetings, digital MEMS microphone array, time difference of arrival (TDOA), superdirective beamforming 1
Transfer learning of language-independent end-to-end ASR with language model fusion
This work explores better adaptation methods to low-resource languages using
an external language model (LM) under the framework of transfer learning. We
first build a language-independent ASR system in a unified sequence-to-sequence
(S2S) architecture with a shared vocabulary among all languages. During
adaptation, we perform LM fusion transfer, where an external LM is integrated
into the decoder network of the attention-based S2S model in the whole
adaptation stage, to effectively incorporate linguistic context of the target
language. We also investigate various seed models for transfer learning.
Experimental evaluations using the IARPA BABEL data set show that LM fusion
transfer improves performances on all target five languages compared with
simple transfer learning when the external text data is available. Our final
system drastically reduces the performance gap from the hybrid systems.Comment: Accepted at ICASSP201
On the Choice of Modeling Unit for Sequence-to-Sequence Speech Recognition
In conventional speech recognition, phoneme-based models outperform
grapheme-based models for non-phonetic languages such as English. The
performance gap between the two typically reduces as the amount of training
data is increased. In this work, we examine the impact of the choice of
modeling unit for attention-based encoder-decoder models. We conduct
experiments on the LibriSpeech 100hr, 460hr, and 960hr tasks, using various
target units (phoneme, grapheme, and word-piece); across all tasks, we find
that grapheme or word-piece models consistently outperform phoneme-based
models, even though they are evaluated without a lexicon or an external
language model. We also investigate model complementarity: we find that we can
improve WERs by up to 9% relative by rescoring N-best lists generated from a
strong word-piece based baseline with either the phoneme or the grapheme model.
Rescoring an N-best list generated by the phonemic system, however, provides
limited improvements. Further analysis shows that the word-piece-based models
produce more diverse N-best hypotheses, and thus lower oracle WERs, than
phonemic models.Comment: To appear in the proceedings of INTERSPEECH 201
Combining vocal tract length normalization with hierarchial linear transformations
Recent research has demonstrated the effectiveness of vocal tract length normalization (VTLN) as a rapid adaptation technique for statistical parametric speech synthesis. VTLN produces speech with naturalness preferable to that of MLLR-based adaptation techniques, being much closer in quality to that generated by the original av-erage voice model. However with only a single parameter, VTLN captures very few speaker specific characteristics when compared to linear transform based adaptation techniques. This paper pro-poses that the merits of VTLN can be combined with those of linear transform based adaptation in a hierarchial Bayesian frame-work, where VTLN is used as the prior information. A novel tech-nique for propagating the gender information from the VTLN prior through constrained structural maximum a posteriori linear regres-sion (CSMAPLR) adaptation is presented. Experiments show that the resulting transformation has improved speech quality with better naturalness, intelligibility and improved speaker similarity. Index Terms — Statistical parametric speech synthesis, hidden Markov models, speaker adaptation, vocal tract length normaliza-tion, constrained structural maximum a posteriori linear regression 1
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