272 research outputs found
End-to-End Attention-based Large Vocabulary Speech Recognition
Many of the current state-of-the-art Large Vocabulary Continuous Speech
Recognition Systems (LVCSR) are hybrids of neural networks and Hidden Markov
Models (HMMs). Most of these systems contain separate components that deal with
the acoustic modelling, language modelling and sequence decoding. We
investigate a more direct approach in which the HMM is replaced with a
Recurrent Neural Network (RNN) that performs sequence prediction directly at
the character level. Alignment between the input features and the desired
character sequence is learned automatically by an attention mechanism built
into the RNN. For each predicted character, the attention mechanism scans the
input sequence and chooses relevant frames. We propose two methods to speed up
this operation: limiting the scan to a subset of most promising frames and
pooling over time the information contained in neighboring frames, thereby
reducing source sequence length. Integrating an n-gram language model into the
decoding process yields recognition accuracies similar to other HMM-free
RNN-based approaches
Recent advances in LVCSR : A benchmark comparison of performances
Large Vocabulary Continuous Speech Recognition (LVCSR), which is characterized by a high variability of the speech, is the most challenging task in automatic speech recognition (ASR). Believing that the evaluation of ASR systems on relevant and common speech corpora is one of the key factors that help accelerating research, we present, in this paper, a benchmark comparison of the performances of the current state-of-the-art LVCSR systems over different speech recognition tasks. Furthermore, we put objectively into evidence the best performing technologies and the best accuracy achieved so far in each task. The benchmarks have shown that the Deep Neural Networks and Convolutional Neural Networks have proven their efficiency on several LVCSR tasks by outperforming the traditional Hidden Markov Models and Guaussian Mixture Models. They have also shown that despite the satisfying performances in some LVCSR tasks, the problem of large-vocabulary speech recognition is far from being solved in some others, where more research efforts are still needed
Analysis of Unsupervised and Noise-Robust Speaker-Adaptive HMM-Based Speech Synthesis Systems toward a Unified ASR and TTS Framework
For the 2009 Blizzard Challenge we have built an unsupervised version of the HTS-2008 speaker-adaptive HMM-based speech synthesis system for English, and a noise robust version of the systems for Mandarin. They are designed from a multidisciplinary application point of view in that we attempt to integrate the components of the TTS system with other technologies such as ASR. All the average voice models are trained exclusively from recognized, publicly available, ASR databases. Multi-pass LVCSR and confidence scores calculated from confusion network are used for the unsupervised systems, and noisy data recorded in cars or public spaces is used for the noise robust system. We believe the developed systems form solid benchmarks and provide good connections to ASR fields. This paper describes the development of the systems and reports the results and analysis of their evaluation
Very Fast Keyword Spotting System with Real Time Factor below 0.01
In the paper we present an architecture of a keyword spotting (KWS) system
that is based on modern neural networks, yields good performance on various
types of speech data and can run very fast. We focus mainly on the last aspect
and propose optimizations for all the steps required in a KWS design: signal
processing and likelihood computation, Viterbi decoding, spot candidate
detection and confidence calculation. We present time and memory efficient
modelling by bidirectional feedforward sequential memory networks (an
alternative to recurrent nets) either by standard triphones or so called
quasi-monophones, and an entirely forward decoding of speech frames (with
minimal need for look back). Several variants of the proposed scheme are
evaluated on 3 large Czech datasets (broadcast, internet and telephone, 17
hours in total) and their performance is compared by Detection Error Tradeoff
(DET) diagrams and real-time (RT) factors. We demonstrate that the complete
system can run in a single pass with a RT factor close to 0.001 if all
optimizations (including a GPU for likelihood computation) are applied.Comment: 11 pages, 3 figure
Transformer-based Automatic Speech Recognition of Formal and Colloquial Czech in MALACH Project
Czech is a very specific language due to its large differences between the
formal and the colloquial form of speech. While the formal (written) form is
used mainly in official documents, literature, and public speeches, the
colloquial (spoken) form is used widely among people in casual speeches. This
gap introduces serious problems for ASR systems, especially when training or
evaluating ASR models on datasets containing a lot of colloquial speech, such
as the MALACH project. In this paper, we are addressing this problem in the
light of a new paradigm in end-to-end ASR systems -- recently introduced
self-supervised audio Transformers. Specifically, we are investigating the
influence of colloquial speech on the performance of Wav2Vec 2.0 models and
their ability to transcribe colloquial speech directly into formal transcripts.
We are presenting results with both formal and colloquial forms in the training
transcripts, language models, and evaluation transcripts.Comment: to be published in Proceedings of TSD 202
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