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TIMIT

About: TIMIT is a research topic. Over the lifetime, 1401 publications have been published within this topic receiving 59888 citations. The topic is also known as: TIMIT Acoustic-Phonetic Continuous Speech Corpus.


Papers
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Proceedings ArticleDOI
19 Apr 2009
TL;DR: In this paper, an instantaneous speech rhythm estimator is introduced to predict possible regions where syllable nuclei can appear, and a simple slope based peak counting algorithm is used to get the exact location of each syllable nucleus.
Abstract: In this paper, we present a novel speech-rhythm-guided syllable-nuclei location detection algorithm. As a departure from conventional methods, we introduce an instantaneous speech rhythm estimator to predict possible regions where syllable nuclei can appear. Within a possible region, a simple slope based peak counting algorithm is used to get the exact location of each syllable nucleus. We verify the correctness of our method by investigating the syllable nuclei interval distribution in TIMIT dataset, and evaluate the performance by comparing with a state-of-the-art syllable nuclei based speech rate detection approach.

42 citations

Posted Content
TL;DR: In this paper, the authors proposed an improved version of PASE for robust speech recognition in noisy and reverberant environments, called PASE+, which employs an online speech distortion module, that contaminates the input signals with a variety of random disturbances.
Abstract: Despite the growing interest in unsupervised learning, extracting meaningful knowledge from unlabelled audio remains an open challenge. To take a step in this direction, we recently proposed a problem-agnostic speech encoder (PASE), that combines a convolutional encoder followed by multiple neural networks, called workers, tasked to solve self-supervised problems (i.e., ones that do not require manual annotations as ground truth). PASE was shown to capture relevant speech information, including speaker voice-print and phonemes. This paper proposes PASE+, an improved version of PASE for robust speech recognition in noisy and reverberant environments. To this end, we employ an online speech distortion module, that contaminates the input signals with a variety of random disturbances. We then propose a revised encoder that better learns short- and long-term speech dynamics with an efficient combination of recurrent and convolutional networks. Finally, we refine the set of workers used in self-supervision to encourage better cooperation. Results on TIMIT, DIRHA and CHiME-5 show that PASE+ significantly outperforms both the previous version of PASE as well as common acoustic features. Interestingly, PASE+ learns transferable representations suitable for highly mismatched acoustic conditions.

42 citations

Posted Content
TL;DR: The performance of a recurrent neural network is compared with the best results published so far on phoneme recognition in the TIMIT database and a single recurrent network is applied to the same task.
Abstract: We compare the performance of a recurrent neural network with the best results published so far on phoneme recognition in the TIMIT database. These published results have been obtained with a combination of classifiers. However, in this paper we apply a single recurrent neural network to the same task. Our recurrent neural network attains an error rate of 24.6%. This result is not significantly different from that obtained by the other best methods, but they rely on a combination of classifiers for achieving comparable performance.

42 citations

Book ChapterDOI
18 Aug 2002
TL;DR: Two approaches in using HMMs (hidden Markov models) to convert audio signals to a sequence of visemes are compared and it is found that the error rates can be reduced to 20.5% and 13.9%, respectably.
Abstract: We describe audio-to-visual conversion techniques for efficient multimedia communications. The audio signals are automatically converted to visual images of mouth shape. The visual speech can be represented as a sequence of visemes, which are the generic face images corresponding to particular sounds. Visual images synchronized with audio signals can provide user-friendly interface for man machine interactions. Also, it can be used to help the people with impaired-hearing. We use HMMs (hidden Markov models) to convert audio signals to a sequence of visemes. In this paper, we compare two approaches in using HMMs. In the first approach, an HMM is trained for each viseme, and the audio signals are directly recognized as a sequence of visemes. In the second approach, each phoneme is modeled with an HMM, and a general phoneme recognizer is utilized to produce a phoneme sequence from the audio signals. The phoneme sequence is then converted to a viseme sequence. We implemented the two approaches and tested them on the TIMIT speech corpus. The viseme recognizer shows 33.9% error rate, and the phoneme-based approach exhibits 29.7% viseme recognition error rate. When similar viseme classes are merged, we have found that the error rates can be reduced to 20.5% and 13.9%, respectably.

42 citations

Posted Content
TL;DR: This paper proposed a stochastic recurrent model, where each step in the sequence is associated with a latent variable that is used to condition the recurrent dynamics for future steps, and training is performed with amortized variational inference where the approximate posterior is augmented with a RNN that runs backward through the sequence.
Abstract: Many efforts have been devoted to training generative latent variable models with autoregressive decoders, such as recurrent neural networks (RNN). Stochastic recurrent models have been successful in capturing the variability observed in natural sequential data such as speech. We unify successful ideas from recently proposed architectures into a stochastic recurrent model: each step in the sequence is associated with a latent variable that is used to condition the recurrent dynamics for future steps. Training is performed with amortized variational inference where the approximate posterior is augmented with a RNN that runs backward through the sequence. In addition to maximizing the variational lower bound, we ease training of the latent variables by adding an auxiliary cost which forces them to reconstruct the state of the backward recurrent network. This provides the latent variables with a task-independent objective that enhances the performance of the overall model. We found this strategy to perform better than alternative approaches such as KL annealing. Although being conceptually simple, our model achieves state-of-the-art results on standard speech benchmarks such as TIMIT and Blizzard and competitive performance on sequential MNIST. Finally, we apply our model to language modeling on the IMDB dataset where the auxiliary cost helps in learning interpretable latent variables. Source Code: \url{this https URL}

42 citations


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Performance
Metrics
No. of papers in the topic in previous years
YearPapers
202324
202262
202167
202086
201977
201895