Jump to Content
Hank Liao

Hank Liao

Hank Liao is a researcher in the speech recognition group with interests in acoustic modeling and large vocabulary speech recognition. Hank Liao obtained a BASc in Systems Design Engineering from the University of Waterloo in 1999. In 2002 he completed an MPhil from the University of Cambridge. In 2007 he submitted his dissertation on "Uncertainty Decoding for Noise Robust Speech Recognition" to the Cambridge University Engineering Department, supervised by Dr. Mark Gales, and sponsored by Toshiba Research Europe. He has published research on and deployed automatic captioning of YouTube videos, Google Voice's voicemail transcription, and Voice Search speech recognition systems.
Authored Publications
Google Publications
Other Publications
Sort By
  • Title
  • Title, descending
  • Year
  • Year, descending
    USM-SCD: USM-Based Multilingual Speaker Change Detection
    Yongqiang Wang
    Jason Pelecanos
    Yu Zhang
    Yiling Huang
    Han Lu
    ICASSP 2024 - 2024 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pp. 11801-11805
    Preview abstract We introduce a multilingual speaker change detection model (USM- SCD) that can simultaneously detect speaker turns and perform ASR for 96 languages. This model is adapted from a speech foundation model trained on a large quantity of supervised and unsupervised data, demonstrating the utility of fine-tuning from a large generic foundation model for a downstream task. We analyze the performance of this multilingual speaker change detection model through a series of ablation studies. We show that the USM-SCD model can achieve more than 75% average speaker change detection F1 score across a test set that consists of data from 96 languages. On American English, the USM-SCD model can achieve an 85.8% speaker change detection F1 score across various public and internal test sets, beating the previous monolingual baseline model by 21% relative. We also show that we only need to fine-tune one-quarter of the trainable model parameters to achieve the best model performance. The USM-SCD model exhibits state-of-the-art ASR quality compared with a strong public ASR baseline, making it suitable to handle both tasks with negligible additional computational cost. View details
    Preview abstract It has been shown that learning audiovisual features can lead to improved speech recognition performance over audio-only features, especially for noisy speech. However, in many common applications, the visual features are partially or entirely missing, e.g.~the speaker might move off screen. Multi-modal models need to be robust: missing video frames should not degrade the performance of an audiovisual model to be worse than that of a single-modality audio-only model. While there have been many attempts at building robust models, there is little consensus on how robustness should be evaluated. To address this, we introduce a framework that allows claims about robustness to be evaluated in a precise and testable way. We also conduct a systematic empirical study of the robustness of common audiovisual speech recognition architectures on a range of acoustic noise conditions and test suites. Finally, we show that an architecture-agnostic solution based on cascades can consistently achieve robustness to missing video, even in settings where existing techniques for robustness like dropout fall short. View details
    Preview abstract Traditionally, audio-visual automatic speech recognition has been studied under the assumption that the speaking face on the visual signal is the face matching the audio. However, in a more realistic setting, when multiple faces are potentially on screen one needs to decide which face to feed to the A/V ASR system. The present work takes the recent progress of A/V ASR one step further and considers the scenario where multiple people are simultaneously on screen (multi-person A/V ASR). We propose a fully differentiable A/V ASR model that is able to handle multiple face tracks in a video. Instead of relying on two separate models for speaker face selection and audio-visual ASR on a single face track, we introduce an attention layer to the ASR encoder that is able to soft-select the appropriate face video track. Experiments carried out on an A/V system trained on over 30k hours of YouTube videos illustrate that the proposed approach can automatically select the proper face tracks with minor WER degradation compared to an oracle selection of the speaking face while still showing benefits of employing the visual signal instead of the audio alone. View details
    Basi Garcia
    Brendan Shillingford
    Yannis Assael
    Proceedings of IEEE Automatic Speech Recognition and Understanding Workshop (2019)
    Preview abstract This work presents a large-scale audio-visual speech recognition system based on a recurrent neural network transducer (RNN-T) architecture. To support the development of such a system, we built a large audio-visual (AV) dataset of segmented utterances extracted from YouTube public videos, leading to 31k hours of audio-visual training content. The performance of an audio-only, visual-only, and audio-visual system are compared on two large-vocabulary test sets: an internal set of YouTube utterances (YouTube-AV-Dev-18) and the publicly available TED-LRS3 set. To highlight the contribution of the visual modality, we also evaluated the performance of our system on the YouTube-AV-Dev-18 set artificially corrupted with additive background noise and overlapping speech. To the best of our knowledge, our system significantly improves the state-of-the-art on the TED-LRS3 set. View details
    Preview abstract Multilingual training has proven to improve acoustic modeling performance by sharing and transferring knowledge in modeling different languages. Knowledge sharing is usually achieved by using common lower-level layers for different languages in a deep neural network. Recently, the domain adversarial network was proposed to reduce domain mismatch of training data and learn domain-invariant features. It is thus worth exploring whether adversarial training can further promote knowledge sharing in multilingual models. In this work, we apply the domain adversarial network to encourage the shared layers of a multilingual model to learn language-invariant features. Bidirectional Long Short-Term Memory (LSTM) recurrent neural networks (RNN) are used as building blocks. We show that shared layers learned this way contain less language identification information and lead to better acoustic modeling performance. In an automatic speech recognition task for seven languages, the resultant acoustic model improves the word error rate (WER) of the multilingual model by a relative 4% on average, and the monolingual models by 10%. View details
    Preview abstract Recurrent neural network language models (RNNLM) and Long Short Term Memory (LSTM) LMs, a variant of RNN LMs, have been shown to outperform traditional N-gram LMs on speech recognition tasks. However, these models are computationally more expensive than N-gram LMs for decoding, and thus, challenging to integrate into speech recognizers. Recent research has proposed the use of lattice-rescoring algorithms using RNNLMs and LSTMLMs as an efficient strategy to integrate these models into a speech recognition system. In this paper, we evaluate existing lattice rescoring algorithms along with a few of our own novel variants on a Youtube speech recognition task. Lattice rescoring using LSTMLMs reduces the word error rate (WER) for this task by about 6\% relative to the WER obtained using an N-gram LM. View details
    Preview abstract We present a new algorithm for efficiently training n-gram language models on uncertain data, and illustrate its use for semi-supervised language model adaptation. We compute the probability that an n-gram occurs k times in the sample of uncertain data, and use the resulting histograms to derive a generalized Katz backoff model. We compare semi-supervised adaptation of language models for YouTube video speech recognition in two conditions: when using full lattices with our new algorithm versus just the 1-best output from the baseline speech recognizer. Unlike 1-best methods, the new algorithm provides models that yield solid improvements over the baseline on the full test set, and, further, achieves these gains without hurting performance on any of the set of channels. We show that channels with the most data yielded the largest gains. The algorithm was implemented via a new semiring in the OpenFst library and will be released as part of the OpenGrm ngram library. View details
    Preview abstract We present results that show it is possible to build a competitive, greatly simplified, large vocabulary continuous speech recognition system with whole words as acoustic units. We model the output vocabulary of about 100,000 words directly using deep bi-directional LSTM RNNs with CTC loss. The model is trained on 125,000 hours of semi-supervised acoustic training data, which enables us to alleviate the data sparsity problem for word models. We show that the CTC word models work very well as an end-to-end all-neural speech recognition model without the use of traditional context-dependent sub-word phone units that require a pronunciation lexicon, and without any language model removing the need to decode. We demonstrate that the CTC word models perform better than a strong, more complex, state-of-the-art baseline with sub-word units. View details
    Preview abstract Recently, Google launched YouTube Kids, a mobile application for children, that uses a speech recognizer built specifically for recognizing children’s speech. In this paper we present techniques we explored to build such a system. We describe the use of a neural network classifier to identify matched acoustic training data, filtering data for language modeling to reduce the chance of producing offensive results. We also compare long short-term memory (LSTM) recurrent networks to convolutional, LSTM, deep neural networks (CLDNN). We found that a CLDNN acoustic model outperforms an LSTM across a variety of different conditions, but does not specifically model child speech relatively better than adult. Overall, these findings allow us to build a successful, state-of-the-art large vocabulary speech recognizer for both children and adults. View details
    GMM-Free DNN Training
    Proceedings of the International Conference on Acoustics,Speech and Signal Processing (2014)
    Speaker Adaptation of Context Dependent Deep Neural Networks
    International Conference of Acoustics, Speech, and Signal Processing. (2013)
    Haptic Voice Recognition Grand Challenge
    K. Sim
    S. Zhao
    K. Yu
    14th ACM International Conference on Multimodal Interaction. (2012)
    Preview abstract In large vocabulary continuous speech recognition, decision trees are widely used to cluster triphone states. In addition to commonly used phonetically based questions, others have proposed additional questions such as phone position within word or syllable. This paper examines using the word or syllable context itself as a feature in the decision tree, providing an elegant way of introducing word- or syllable-specific models into the system. Positive results are reported on two state-of-the-art systems: voicemail transcription and a search by voice tasks across a variety of acoustic model and training set sizes. View details
    An Audio Indexing System for Election Video Material
    Christopher Alberti
    Ari Bezman
    Anastassia Drofa
    Ted Power
    Arnaud Sahuguet
    Maria Shugrina
    Proceedings of ICASSP (2009), pp. 4873-4876
    Preview abstract In the 2008 presidential election race in the United States, the prospective candidates made extensive use of YouTube to post video material. We developed a scalable system that transcribes this material and makes the content searchable (by indexing the meta-data and transcripts of the videos) and allows the user to navigate through the video material based on content. The system is available as an iGoogle gadget as well as a Labs product. Given the large exposure, special emphasis was put on the scalability and reliability of the system. This paper describes the design and implementation of this system. View details
    Issues with uncertainty decoding for noise robust automatic speech recognition
    M.J.F. Gales
    Speech Communication, vol. 50 (2008), pp. 265-277
    Adaptive Training with Joint Uncertainty Decoding for Robust Recognition of Noise Data
    M.J.F. Gales
    ICASSP (2007)
    Uncertainty Decoding for Noise Robust Speech Recognition
    Ph.D. Thesis, University of Cambridge (2007)
    Issues with Uncertainty Decoding for Noise Robust Speech Recognition. In Proc. Interspeech
    M.J.F. Gales
    Interspeech (2006)
    Joint Uncertainty Decoding for Robust Large Vocabulary Speech Recognition
    M.J.F. Gales
    University of Cambridge (2006)
    Issues with Uncertainty Decoding for Noise Robust Speech Recognition
    M.J.F. Gales
    University of Cambridge (2006)
    Joint Uncertainty Decoding for Noise Robust Speech Recognition. In Proc. Interspeech
    M.J.F. Gales
    Interspeech (2005)
    Uncertainty Decoding for Noise Robust Speech Recognition
    M.J.F. Gales
    University of Cambridge (2004)