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 conversational speech recognition


Building competitive direct acoustics-to-word models for English conversational speech recognition

arXiv.org Machine Learning

Direct acoustics-to-word (A2W) models in the end-to-end paradigm have received increasing attention compared to conventional sub-word based automatic speech recognition models using phones, characters, or context-dependent hidden Markov model states. This is because A2W models recognize words from speech without any decoder, pronunciation lexicon, or externally-trained language model, making training and decoding with such models simple. Prior work has shown that A2W models require orders of magnitude more training data in order to perform comparably to conventional models. Our work also showed this accuracy gap when using the English Switchboard-Fisher data set. This paper describes a recipe to train an A2W model that closes this gap and is at-par with state-of-the-art sub-word based models. We achieve a word error rate of 8.8%/13.9% on the Hub5-2000 Switchboard/CallHome test sets without any decoder or language model. We find that model initialization, training data order, and regularization have the most impact on the A2W model performance. Next, we present a joint word-character A2W model that learns to first spell the word and then recognize it. This model provides a rich output to the user instead of simple word hypotheses, making it especially useful in the case of words unseen or rarely-seen during training.


Achieving human parity in conversational speech recognition

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The headline story here is that for the first time a system has been developed that exceeds human performance in one of the most difficult of all human speech recognition tasks: natural conversations held over the telephone. This is known as conversational telephone speech, or CTS. The reference datasets for this task are the Switchboard and Fisher data collections from the 1990s and early 2000s. The apocryphal story here is that human performance on the task is about 4% error rate. But no-one can quite pin down where that 4% number comes from.


Microsoft researchers achieve a historic milestone and reach human parity in conversational speech recognition - The Fire Hose

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Microsoft has made a major breakthrough in speech recognition, creating a technology that understands a conversation as well as a person does. In a paper published Monday, a team of researchers and engineers in Microsoft Artificial Intelligence and Research reported a speech recognition system that makes the same or fewer errors than professional transcriptionists. The researchers reported a word error rate (WER) of 5.9 percent, down from the 6.3 percent WER the team reported just last month. The 5.9 percent error rate is about equal to that of people who were asked to transcribe the same conversation, and it's the lowest ever recorded against the industry standard Switchboard speech recognition task.


Historic Achievement: Microsoft researchers reach human parity in conversational speech recognition - Next at Microsoft

#artificialintelligence

Microsoft has made a major breakthrough in speech recognition, creating a technology that recognizes the words in a conversation as well as a person does. In a paper published Monday, a team of researchers and engineers in Microsoft Artificial Intelligence and Research reported a speech recognition system that makes the same or fewer errors than professional transcriptionists. The researchers reported a word error rate (WER) of 5.9 percent, down from the 6.3 percent WER the team reported just last month. The 5.9 percent error rate is about equal to that of people who were asked to transcribe the same conversation, and it's the lowest ever recorded against the industry standard Switchboard speech recognition task. "We've reached human parity," said Xuedong Huang, the company's chief speech scientist.



Recent Advances in Conversational Speech Recognition

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Our second model, called very deep convolutional neural net (or CNN), has its origins in image classification [4]. Speech can be viewed as an image if we consider the spectral representation of the audio signal with the two dimensions being time and frequency. As opposed to the classic CNN architectures employed in our previous system [5] that have only one or two convolutional layers with large (typically 9-by-9) kernels, our very deep CNN [6] has up to ten convolutional layers with small 3-by-3 kernels which preserve the dimensionality of the input. By stacking many of these convolutional layers with Rectified Linear Units nonlinearities before pooling layers, the same receptive field is created with less parameters and more nonlinearity. These two models which differ radically in architecture and input representation show good complementarity and their combination leads to additional gains over the best individual model.