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Zhong Meng

23 accepted papers

2024

A Comparison of Parameter-Efficient ASR Domain Adaptation Methods for Universal Speech and Language Models

ICASSP 2024accepted

A recent paradigm shift in artificial intelligence has seen the rise of foundation models, such as the large language models and the universal speech models. With billions of model parameters and trained with a wide range of data, these foundation models are expected to have a better generalization…

Cited by 0SourceScholar
2024

Augmenting Conformers With Structured State-Space Sequence Models For Online Speech Recognition

ICASSP 2024accepted

Online speech recognition, where the model only accesses context to the left, is an important and challenging use case for ASR systems. In this work, we investigate augmenting neural encoders for online ASR by incorporating structured state-space sequence models (S4), a family of models that provide…

Cited by 0SourceScholar
2024

Deferred NAM: Low-latency Top-K Context Injection via Deferred Context Encoding for Non-Streaming ASR

NAACL 2024industry

Contextual biasing enables speech recognizers to transcribe important phrases in the speaker’s context, such as contact names, even if they are rare in, or absent from, the training data. Attention-based biasing is a leading approach which allows for full end-to-end cotraining of the recognizer and…

Cited by 2SourcePDFScholar
2024

Extreme Encoder Output Frame Rate Reduction: Improving Computational Latencies of Large End-to-End Models

ICASSP 2024accepted

The accuracy of end-to-end (E2E) automatic speech recognition (ASR) models continues to improve as they are scaled to larger sizes, with some now reaching billions of parameters. Widespread deployment and adoption of these models, however, requires computationally efficient strategies for decoding.…

Cited by 0SourceScholar
2024

Massive End-to-end Speech Recognition Models with Time Reduction

NAACL 2024long

We investigate massive end-to-end automatic speech recognition (ASR) models with efficiency improvements achieved by time reduction. The encoders of our models use the neural architecture of Google’s universal speech model (USM), with additional funnel pooling layers to significantly reduce the fram…

Cited by 2SourcePDFScholar
2023

JEIT: Joint End-to-End Model and Internal Language Model Training for Speech Recognition

ICASSP 2023accepted

We propose JEIT, a joint end-to-end (E2E) model and internal language model (ILM) training method to inject large-scale unpaired text into ILM during E2E training which improves rare-word speech recognition. With JEIT, the E2E model computes an E2E loss on audio-transcript pairs while its ILM estima…

Cited by 0SourceScholar
2022

Continuous Speech Separation with Recurrent Selective Attention Network

ICASSP 2022accepted

While permutation invariant training (PIT) based continuous speech separation (CSS) significantly improves the conversation transcription accuracy, it often suffers from speech leakages and failures in separation at "hot spot" regions because it has a fixed number of output channels. In this paper,…

Cited by 0SourceScholar
2022

Factorized Neural Transducer for Efficient Language Model Adaptation

ICASSP 2022accepted

In recent years, end-to-end (E2E) based automatic speech recognition (ASR) systems have achieved great success due to their simplicity and promising performance. Neural Transducer based models are increasingly popular in streaming E2E based ASR systems and have been reported to outperform the tradit…

Cited by 0SourceScholar
2022

Transcribe-to-Diarize: Neural Speaker Diarization for Unlimited Number of Speakers Using End-to-End Speaker-Attributed ASR

ICASSP 2022accepted

This paper presents Transcribe-to-Diarize, a new approach for neural speaker diarization that uses an end-to-end (E2E) speaker-attributed automatic speech recognition (SA-ASR). The E2E SA-ASR is a joint model that was recently proposed for speaker counting, multi-talker speech recognition, and speak…

Cited by 0SourceScholar
2021

Hypothesis Stitcher for End-to-End Speaker-Attributed ASR on Long-Form Multi-Talker Recordings

ICASSP 2021accepted

An end-to-end (E2E) speaker-attributed automatic speech recognition (SA-ASR) model was proposed recently to jointly perform speaker counting, speech recognition and speaker identification. The model achieved a low speaker-attributed word error rate (SA-WER) for monaural overlapped speech comprising…

Cited by 0SourceScholar
2021

Internal Language Model Training for Domain-Adaptive End-To-End Speech Recognition

ICASSP 2021accepted

The efficacy of external language model (LM) integration with existing end-to-end (E2E) automatic speech recognition (ASR) systems can be improved significantly using the internal language model estimation (ILME) method [1]. In this method, the internal LM score is subtracted from the score obtained…

Cited by 0SourceScholar
2021

Minimum Bayes Risk Training for End-to-End Speaker-Attributed ASR

ICASSP 2021accepted

Recently, an end-to-end speaker-attributed automatic speech recognition (E2E SA-ASR) model was proposed as a joint model of speaker counting, speech recognition and speaker identification for monaural overlapped speech. In the previous study, the model parameters were trained based on the speaker-at…

Cited by 0SourceScholar
2020

Continuous Speech Separation: Dataset and Analysis

ICASSP 2020accepted

This paper describes a dataset and protocols for evaluating continuous speech separation algorithms. Most prior speech separation studies use pre-segmented audio signals, which are typically generated by mixing speech utterances on computers so that they fully overlap. Also, the separation algorithm…

Cited by 0SourceScholar
2020

High-Accuracy and Low-Latency Speech Recognition with Two-Head Contextual Layer Trajectory LSTM Model

ICASSP 2020accepted

While the community keeps promoting end-to-end models over conventional hybrid models, which usually are long short-term memory (LSTM) models trained with a cross entropy criterion followed by a sequence discriminative training criterion, we argue that such conventional hybrid models can still be si…

Cited by 0SourceScholar
2020

L-Vector: Neural Label Embedding for Domain Adaptation

ICASSP 2020accepted

We propose a novel neural label embedding (NLE) scheme for the domain adaptation of a deep neural network (DNN) acoustic model with unpaired data samples from source and target domains. With NLE method, we distill the knowledge from a powerful source-domain DNN into a dictionary of label embeddings,…

Cited by 0SourceScholar
2019

Adversarial Speaker Adaptation

ICASSP 2019accepted

We propose a novel adversarial speaker adaptation (ASA) scheme, in which adversarial learning is applied to regularize the distribution of deep hidden features in a speaker-dependent (SD) deep neural network (DNN) acoustic model to be close to that of a fixed speaker-independent (SI) DNN acoustic mo…

Cited by 0SourceScholar
2018

Adversarial Teacher-Student Learning for Unsupervised Domain Adaptation

ICASSP 2018accepted

The teacher-student (T/S) learning has been shown effective in unsupervised domain adaptation [1]. It is a form of transfer learning, not in terms of the transfer of recognition decisions, but the knowledge of posteriori probabilities in the source domain as evaluated by the teacher model. It learns…

Cited by 0SourceScholar
2018

Speaker-Invariant Training Via Adversarial Learning

ICASSP 2018accepted

We propose a novel adversarial multi-task learning scheme, aiming at actively curtailing the inter-talker feature variability while maximizing its senone discriminability so as to enhance the performance of a deep neural network (DNN) based ASR system. We call the scheme speaker-invariant training (…

Cited by 0SourceScholar
2017

Deep long short-term memory adaptive beamforming networks for multichannel robust speech recognition

ICASSP 2017accepted

Far-field speech recognition in noisy and reverberant conditions remains a challenging problem despite recent deep learning breakthroughs. This problem is commonly addressed by acquiring a speech signal from multiple microphones and performing beamforming over them. In this paper, we propose to use…

Cited by 0SourceScholar