William Ravenscroft

dblp:318/2725 · DBLP profile ↗
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7ranked-venue papers
5as first author
7since 2021 · last 2025
0000-0002-0780-3303ORCID · verified

Domains — the database's venue-derived domains; a paper can count in several

Graphics, computer vision, multimedia, augmented reality and games · 7 · 5 first-author · 7 since 2021Artificial intelligence and machine learning · 3 · 3 first-author · 3 since 2021
YearPublicationVenuePosition
2025 Whilter: A Whisper-based Data Filter for "In-the-Wild" Speech Corpora Using Utterance-level Multi-Task Classification
William Ravenscroft, George Close, Kit Bower-Morris, Jamie Stacey, Dmitry Sityaev, Kris Y. Hong
INTERSPEECH1
2024 Multi-CMGAN+/+: Leveraging Multi-Objective Speech Quality Metric Prediction for Speech Enhancement
abstract
Neural network based approaches to speech enhancement have shown to be particularly powerful, being able to leverage a data-driven approach to result in a significant performance gain versus other approaches. Such approaches are reliant on artificially created labelled training data such that the neural model can be trained using intrusive loss functions which compare the output of the model with clean reference speech. Performance of such systems when enhancing real-world audio often suffers relative to their performance on simulated test data. In this work, a non-intrusive multi-metric prediction approach is introduced, wherein a model trained on artificial labelled data using inference of an adversarially trained metric prediction neural network. The proposed approach shows improved performance versus state-of-the-art systems on the recent CHiME-7 challenge unsupervised domain adaptation speech enhancement (UDASE) task evaluation sets. Index Terms: speech enhancement, model generalisation, generative adversarial networks, conformer, metric prediction
George Close, William Ravenscroft, Thomas Hain, Stefan Goetze
ICASSP2
2024 Combining Conformer and Dual-Path-Transformer Networks for Single Channel Noisy Reverberant Speech Separation
abstract
Separation of overlapping speakers remains an active area of speech technology research. Many deep neural network (DNN) separation models propose modelling local and global temporal context separately using alternating DNN layers. Two such models are SepFormer and TD-Conformer. The largest configurations of each have comparable computational cost and similar performance; with SepFormer performing better on anechoic data and TD-Conformer yielding better results on noisy reverberant data. This work combines these two model types to gain insights into how their computational characteristics affect their performance. The generalization benefits of the larger model size of the conformer layers are demonstrated both on the WHAMR and the out-of-domain far-field evaluation set MC-WSJ-AV across a number of evaluation metrics. The proposed model is able to achieve 22.1 dB and 14.7 dB average scale-invariant signal-to-distortion ratio (SISDR) improvement when trained and evaluated on WSJ0-2Mix and WHAMR, respectively. The model trained using WHAMR is able to achieve 4.3 dB average SISDR improvement on the out-of-domain MC-WSJ-AV dataset.
William Ravenscroft, Stefan Goetze, Thomas Hain
ICASSP1
2024 Transcription-Free Fine-Tuning of Speech Separation Models for Noisy and Reverberant Multi-Speaker Automatic Speech Recognition
abstract
One solution to automatic speech recognition (ASR) of overlapping speakers is to separate speech and then perform ASR on the separated signals.Commonly, the separator produces artefacts which often degrade ASR performance.Addressing this issue typically requires reference transcriptions to jointly train the separation and ASR networks.This is often not viable for training on real-world in-domain audio where reference transcript information is not always available.This paper proposes a transcription-free method for joint training using only audio signals.The proposed method uses embedding differences of pre-trained ASR encoders as a loss with a proposed modification to permutation invariant training (PIT) called guided PIT (GPIT).The method achieves a 6.4% improvement in word error rate (WER) measures over a signal-level loss and also shows enhancement improvements in perceptual measures such as short-time objective intelligibility (STOI).
William Ravenscroft, George Close, Stefan Goetze, Thomas Hain, Mohammad Soleymanpour, Anurag Chowdhury, Mark C. Fuhs
INTERSPEECH1
2023 On Time Domain Conformer Models for Monaural Speech Separation in Noisy Reverberant Acoustic Environments
abstract
Speech separation remains an important topic for multispeaker technology researchers. Convolution augmented transformers (conformers) have performed well for many speech processing tasks but have been under-researched for speech separation. Most recent state-of-the-art (SOTA) separation models have been time-domain audio separation networks (TasNets). A number of successful models have made use of dual-path (DP) networks which sequentially process local and global information. Time domain conformers (TD-Conformers) are an analogue of the DP approach in that they also process local and global context sequentially but have a different time complexity function. It is shown that for realistic shorter signal lengths, conformers are more efficient when controlling for feature dimension. Subsampling layers are proposed to further improve computational efficiency. The best TD-Conformer achieves $14.6 \mathrm{~dB}$ and $21.2 \mathrm{~dB}$ SISDR improvement on the WHAMR and WSJO2Mix benchmarks, respectively.
William Ravenscroft, Stefan Goetze, Thomas Hain
ASRU1
2023 Perceive and Predict: Self-Supervised Speech Representation Based Loss Functions for Speech Enhancement
abstract
Recent work in the domain of speech enhancement has explored the use of self-supervised speech representations to aid in the training of neural speech enhancement models. However, much of this work focuses on using the deepest or final outputs of self supervised speech representation models, rather than the earlier feature encodings. The use of self supervised representations in such a way is often not fully motivated. In this work it is shown that the distance between the feature encodings of clean and noisy speech correlate strongly with psychoacoustically motivated measures of speech quality and intelligibility, as well as with human Mean Opinion Score (MOS) ratings. Experiments using this distance as a loss function are performed and improved performance over the use of STFT spectrogram distance based loss as well as other common loss functions from speech enhancement literature is demonstrated using objective measures such as perceptual evaluation of speech quality (PESQ) and short-time objective intelligibility (STOI).
George Close, William Ravenscroft, Thomas Hain, Stefan Goetze
ICASSP2
2023 Deformable Temporal Convolutional Networks for Monaural Noisy Reverberant Speech Separation
abstract
Speech separation models are used for isolating individual speakers in many speech processing applications. Deep learning models have been shown to lead to state-of-the-art (SOTA) results on a number of speech separation benchmarks. One such class of models known as temporal convolutional networks (TCNs) has shown promising results for speech separation tasks. A limitation of these models is that they have a fixed receptive field (RF). Recent research in speech dereverberation has shown that the optimal RF of a TCN varies with the reverberation characteristics of the speech signal. In this work deformable convolution is proposed as a solution to allow TCN models to have dynamic RFs that can adapt to various reverberation times for reverberant speech separation. The proposed models are capable of achieving an 11.1 dB average scale-invariant signal-to-distortion ratio (SISDR) improvement over the input signal on the WHAMR benchmark. A relatively small deformable TCN model of 1.3M parameters is proposed which gives comparable separation performance to larger and more computationally complex models.
William Ravenscroft, Stefan Goetze, Thomas Hain
ICASSP1