149 research outputs found
Gaze modulated disambiguation technique for gesture control in 3D virtual objects selection
© 2017 IEEE. Inputs with multimodal information provide more natural ways to interact with virtual 3D environment. An emerging technique that integrates gaze modulated pointing with mid-air gesture control enables fast target acquisition and rich control expressions. The performance of this technique relies on the eye tracking accuracy which is not comparable with the traditional pointing techniques (e.g., mouse) yet. This will cause troubles when fine grainy interactions are required, such as selecting in a dense virtual scene where proximity and occlusion are prone to occur. This paper proposes a coarse-to-fine solution to compensate the degradation introduced by eye tracking inaccuracy using a gaze cone to detect ambiguity and then a gaze probe for decluttering. It is tested in a comparative experiment which involves 12 participants with 3240 runs. The results show that the proposed technique enhanced the selection accuracy and user experience but it is still with a potential to be improved in efficiency. This study contributes to providing a robust multimodal interface design supported by both eye tracking and mid-air gesture control
OCHID-Fi: Occlusion-Robust Hand Pose Estimation in 3D via RF-Vision
Hand Pose Estimation (HPE) is crucial to many applications, but conventional
cameras-based CM-HPE methods are completely subject to Line-of-Sight (LoS), as
cameras cannot capture occluded objects. In this paper, we propose to exploit
Radio-Frequency-Vision (RF-vision) capable of bypassing obstacles for achieving
occluded HPE, and we introduce OCHID-Fi as the first RF-HPE method with 3D pose
estimation capability. OCHID-Fi employs wideband RF sensors widely available on
smart devices (e.g., iPhones) to probe 3D human hand pose and extract their
skeletons behind obstacles. To overcome the challenge in labeling RF imaging
given its human incomprehensible nature, OCHID-Fi employs a cross-modality and
cross-domain training process. It uses a pre-trained CM-HPE network and a
synchronized CM/RF dataset, to guide the training of its complex-valued RF-HPE
network under LoS conditions. It further transfers knowledge learned from
labeled LoS domain to unlabeled occluded domain via adversarial learning,
enabling OCHID-Fi to generalize to unseen occluded scenarios. Experimental
results demonstrate the superiority of OCHID-Fi: it achieves comparable
accuracy to CM-HPE under normal conditions while maintaining such accuracy even
in occluded scenarios, with empirical evidence for its generalizability to new
domains.Comment: Accepted to ICCV 202
Two-pass Decoding and Cross-adaptation Based System Combination of End-to-end Conformer and Hybrid TDNN ASR Systems
Fundamental modelling differences between hybrid and end-to-end (E2E)
automatic speech recognition (ASR) systems create large diversity and
complementarity among them. This paper investigates multi-pass rescoring and
cross adaptation based system combination approaches for hybrid TDNN and
Conformer E2E ASR systems. In multi-pass rescoring, state-of-the-art hybrid
LF-MMI trained CNN-TDNN system featuring speed perturbation, SpecAugment and
Bayesian learning hidden unit contributions (LHUC) speaker adaptation was used
to produce initial N-best outputs before being rescored by the speaker adapted
Conformer system using a 2-way cross system score interpolation. In cross
adaptation, the hybrid CNN-TDNN system was adapted to the 1-best output of the
Conformer system or vice versa. Experiments on the 300-hour Switchboard corpus
suggest that the combined systems derived using either of the two system
combination approaches outperformed the individual systems. The best combined
system obtained using multi-pass rescoring produced statistically significant
word error rate (WER) reductions of 2.5% to 3.9% absolute (22.5% to 28.9%
relative) over the stand alone Conformer system on the NIST Hub5'00, Rt03 and
Rt02 evaluation data.Comment: It' s accepted to ISCA 202
Adversarial Data Augmentation Using VAE-GAN for Disordered Speech Recognition
Automatic recognition of disordered speech remains a highly challenging task
to date. The underlying neuro-motor conditions, often compounded with
co-occurring physical disabilities, lead to the difficulty in collecting large
quantities of impaired speech required for ASR system development. This paper
presents novel variational auto-encoder generative adversarial network
(VAE-GAN) based personalized disordered speech augmentation approaches that
simultaneously learn to encode, generate and discriminate synthesized impaired
speech. Separate latent features are derived to learn dysarthric speech
characteristics and phoneme context representations. Self-supervised
pre-trained Wav2vec 2.0 embedding features are also incorporated. Experiments
conducted on the UASpeech corpus suggest the proposed adversarial data
augmentation approach consistently outperformed the baseline speed perturbation
and non-VAE GAN augmentation methods with trained hybrid TDNN and End-to-end
Conformer systems. After LHUC speaker adaptation, the best system using VAE-GAN
based augmentation produced an overall WER of 27.78% on the UASpeech test set
of 16 dysarthric speakers, and the lowest published WER of 57.31% on the subset
of speakers with "Very Low" intelligibility.Comment: Submitted to ICASSP 202
Audio-visual End-to-end Multi-channel Speech Separation, Dereverberation and Recognition
Accurate recognition of cocktail party speech containing overlapping
speakers, noise and reverberation remains a highly challenging task to date.
Motivated by the invariance of visual modality to acoustic signal corruption,
an audio-visual multi-channel speech separation, dereverberation and
recognition approach featuring a full incorporation of visual information into
all system components is proposed in this paper. The efficacy of the video
input is consistently demonstrated in mask-based MVDR speech separation,
DNN-WPE or spectral mapping (SpecM) based speech dereverberation front-end and
Conformer ASR back-end. Audio-visual integrated front-end architectures
performing speech separation and dereverberation in a pipelined or joint
fashion via mask-based WPD are investigated. The error cost mismatch between
the speech enhancement front-end and ASR back-end components is minimized by
end-to-end jointly fine-tuning using either the ASR cost function alone, or its
interpolation with the speech enhancement loss. Experiments were conducted on
the mixture overlapped and reverberant speech data constructed using simulation
or replay of the Oxford LRS2 dataset. The proposed audio-visual multi-channel
speech separation, dereverberation and recognition systems consistently
outperformed the comparable audio-only baseline by 9.1% and 6.2% absolute
(41.7% and 36.0% relative) word error rate (WER) reductions. Consistent speech
enhancement improvements were also obtained on PESQ, STOI and SRMR scores.Comment: IEEE/ACM Transactions on Audio, Speech, and Language Processin
Exploring Self-supervised Pre-trained ASR Models For Dysarthric and Elderly Speech Recognition
Automatic recognition of disordered and elderly speech remains a highly
challenging task to date due to the difficulty in collecting such data in large
quantities. This paper explores a series of approaches to integrate domain
adapted SSL pre-trained models into TDNN and Conformer ASR systems for
dysarthric and elderly speech recognition: a) input feature fusion between
standard acoustic frontends and domain adapted wav2vec2.0 speech
representations; b) frame-level joint decoding of TDNN systems separately
trained using standard acoustic features alone and with additional wav2vec2.0
features; and c) multi-pass decoding involving the TDNN/Conformer system
outputs to be rescored using domain adapted wav2vec2.0 models. In addition,
domain adapted wav2vec2.0 representations are utilized in
acoustic-to-articulatory (A2A) inversion to construct multi-modal dysarthric
and elderly speech recognition systems. Experiments conducted on the UASpeech
dysarthric and DementiaBank Pitt elderly speech corpora suggest TDNN and
Conformer ASR systems integrated domain adapted wav2vec2.0 models consistently
outperform the standalone wav2vec2.0 models by statistically significant WER
reductions of 8.22% and 3.43% absolute (26.71% and 15.88% relative) on the two
tasks respectively. The lowest published WERs of 22.56% (52.53% on very low
intelligibility, 39.09% on unseen words) and 18.17% are obtained on the
UASpeech test set of 16 dysarthric speakers, and the DementiaBank Pitt test set
respectively.Comment: accepted by ICASSP 202
Confidence Score Based Speaker Adaptation of Conformer Speech Recognition Systems
Speaker adaptation techniques provide a powerful solution to customise
automatic speech recognition (ASR) systems for individual users. Practical
application of unsupervised model-based speaker adaptation techniques to data
intensive end-to-end ASR systems is hindered by the scarcity of speaker-level
data and performance sensitivity to transcription errors. To address these
issues, a set of compact and data efficient speaker-dependent (SD) parameter
representations are used to facilitate both speaker adaptive training and
test-time unsupervised speaker adaptation of state-of-the-art Conformer ASR
systems. The sensitivity to supervision quality is reduced using a confidence
score-based selection of the less erroneous subset of speaker-level adaptation
data. Two lightweight confidence score estimation modules are proposed to
produce more reliable confidence scores. The data sparsity issue, which is
exacerbated by data selection, is addressed by modelling the SD parameter
uncertainty using Bayesian learning. Experiments on the benchmark 300-hour
Switchboard and the 233-hour AMI datasets suggest that the proposed confidence
score-based adaptation schemes consistently outperformed the baseline
speaker-independent (SI) Conformer model and conventional non-Bayesian, point
estimate-based adaptation using no speaker data selection. Similar consistent
performance improvements were retained after external Transformer and LSTM
language model rescoring. In particular, on the 300-hour Switchboard corpus,
statistically significant WER reductions of 1.0%, 1.3%, and 1.4% absolute
(9.5%, 10.9%, and 11.3% relative) were obtained over the baseline SI Conformer
on the NIST Hub5'00, RT02, and RT03 evaluation sets respectively. Similar WER
reductions of 2.7% and 3.3% absolute (8.9% and 10.2% relative) were also
obtained on the AMI development and evaluation sets.Comment: IEEE/ACM Transactions on Audio, Speech, and Language Processin
Building High-accuracy Multilingual ASR with Gated Language Experts and Curriculum Training
We propose gated language experts and curriculum training to enhance
multilingual transformer transducer models without requiring language
identification (LID) input from users during inference. Our method incorporates
a gating mechanism and LID loss, enabling transformer experts to learn
language-specific information. By combining gated transformer experts with
shared transformer layers, we construct multilingual transformer blocks and
utilize linear experts to effectively regularize the joint network. The
curriculum training scheme leverages LID to guide the gated experts in
improving their respective language performance. Experimental results on a
bilingual task involving English and Spanish demonstrate significant
improvements, with average relative word error reductions of 12.5% and 7.3%
compared to the baseline bilingual and monolingual models, respectively.
Notably, our method achieves performance comparable to the upper-bound model
trained and inferred with oracle LID. Extending our approach to trilingual,
quadrilingual, and pentalingual models reveals similar advantages to those
observed in the bilingual models, highlighting its ease of extension to
multiple languages
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