The RoyalFlush System of Speech Recognition for M2MeT Challenge
- URL: http://arxiv.org/abs/2202.01614v1
- Date: Thu, 3 Feb 2022 14:38:26 GMT
- Title: The RoyalFlush System of Speech Recognition for M2MeT Challenge
- Authors: Shuaishuai Ye, Peiyao Wang, Shunfei Chen, Xinhui Hu, and Xinkang Xu
- Abstract summary: This paper describes our RoyalFlush system for the track of multi-speaker automatic speech recognition (ASR) in the M2MeT challenge.
We adopted the serialized output training (SOT) based multi-speakers ASR system with large-scale simulation data.
Our system got a 12.22% absolute Character Error Rate (CER) reduction on the validation set and 12.11% on the test set.
- Score: 5.863625637354342
- License: http://arxiv.org/licenses/nonexclusive-distrib/1.0/
- Abstract: This paper describes our RoyalFlush system for the track of multi-speaker
automatic speech recognition (ASR) in the M2MeT challenge. We adopted the
serialized output training (SOT) based multi-speakers ASR system with
large-scale simulation data. Firstly, we investigated a set of front-end
methods, including multi-channel weighted predicted error (WPE), beamforming,
speech separation, speech enhancement and so on, to process training,
validation and test sets. But we only selected WPE and beamforming as our
frontend methods according to their experimental results. Secondly, we made
great efforts in the data augmentation for multi-speaker ASR, mainly including
adding noise and reverberation, overlapped speech simulation, multi-channel
speech simulation, speed perturbation, front-end processing, and so on, which
brought us a great performance improvement. Finally, in order to make full use
of the performance complementary of different model architecture, we trained
the standard conformer based joint CTC/Attention (Conformer) and U2++ ASR model
with a bidirectional attention decoder, a modification of Conformer, to fuse
their results. Comparing with the official baseline system, our system got a
12.22% absolute Character Error Rate (CER) reduction on the validation set and
12.11% on the test set.
Related papers
- MLCA-AVSR: Multi-Layer Cross Attention Fusion based Audio-Visual Speech Recognition [62.89464258519723]
We propose a multi-layer cross-attention fusion based AVSR approach that promotes representation of each modality by fusing them at different levels of audio/visual encoders.
Our proposed approach surpasses the first-place system, establishing a new SOTA cpCER of 29.13% on this dataset.
arXiv Detail & Related papers (2024-01-07T08:59:32Z) - Improving Audio-Visual Speech Recognition by Lip-Subword Correlation
Based Visual Pre-training and Cross-Modal Fusion Encoder [58.523884148942166]
We propose two novel techniques to improve audio-visual speech recognition (AVSR) under a pre-training and fine-tuning training framework.
First, we explore the correlation between lip shapes and syllable-level subword units in Mandarin to establish good frame-level syllable boundaries from lip shapes.
Next, we propose an audio-guided cross-modal fusion encoder (CMFE) neural network to utilize main training parameters for multiple cross-modal attention layers.
arXiv Detail & Related papers (2023-08-14T08:19:24Z) - Audio-visual End-to-end Multi-channel Speech Separation, Dereverberation
and Recognition [52.11964238935099]
An audio-visual multi-channel speech separation, dereverberation and recognition approach is proposed in this paper.
Video input is consistently demonstrated in mask-based MVDR speech separation, DNN-WPE or spectral mapping (SpecM) based speech dereverberation front-end.
Experiments were conducted on the mixture overlapped and reverberant speech data constructed using simulation or replay of the Oxford LRS2 dataset.
arXiv Detail & Related papers (2023-07-06T10:50:46Z) - Fully Automated End-to-End Fake Audio Detection [57.78459588263812]
This paper proposes a fully automated end-toend fake audio detection method.
We first use wav2vec pre-trained model to obtain a high-level representation of the speech.
For the network structure, we use a modified version of the differentiable architecture search (DARTS) named light-DARTS.
arXiv Detail & Related papers (2022-08-20T06:46:55Z) - The USTC-Ximalaya system for the ICASSP 2022 multi-channel multi-party
meeting transcription (M2MeT) challenge [43.262531688434215]
We propose two improvements to target-speaker voice activity detection (TS-VAD)
These techniques are designed to handle multi-speaker conversations in real-world meeting scenarios with high speaker-overlap ratios and under heavy reverberant and noisy condition.
arXiv Detail & Related papers (2022-02-10T06:06:48Z) - The Volcspeech system for the ICASSP 2022 multi-channel multi-party
meeting transcription challenge [18.33054364289739]
This paper describes our submission to ICASSP 2022 Multi-channel Multi-party Meeting Transcription (M2MeT) Challenge.
For Track 1, we propose several approaches to empower the clustering-based speaker diarization system.
For Track 2, we develop our system using the Conformer model in a joint CTC-attention architecture.
arXiv Detail & Related papers (2022-02-09T03:38:39Z) - Cross-Modal ASR Post-Processing System for Error Correction and
Utterance Rejection [25.940199825317073]
We propose a cross-modal post-processing system for speech recognizers.
It fuses acoustic features and textual features from different modalities.
It joints a confidence estimator and an error corrector in multi-task learning fashion.
arXiv Detail & Related papers (2022-01-10T12:29:55Z) - Audio-visual Multi-channel Recognition of Overlapped Speech [79.21950701506732]
This paper presents an audio-visual multi-channel overlapped speech recognition system featuring tightly integrated separation front-end and recognition back-end.
Experiments suggest that the proposed multi-channel AVSR system outperforms the baseline audio-only ASR system by up to 6.81% (26.83% relative) and 22.22% (56.87% relative) absolute word error rate (WER) reduction on overlapped speech constructed using either simulation or replaying of the lipreading sentence 2 dataset respectively.
arXiv Detail & Related papers (2020-05-18T10:31:19Z) - You Do Not Need More Data: Improving End-To-End Speech Recognition by
Text-To-Speech Data Augmentation [59.31769998728787]
We build our TTS system on an ASR training database and then extend the data with synthesized speech to train a recognition model.
Our system establishes a competitive result for end-to-end ASR trained on LibriSpeech train-clean-100 set with WER 4.3% for test-clean and 13.5% for test-other.
arXiv Detail & Related papers (2020-05-14T17:24:57Z) - Multiresolution and Multimodal Speech Recognition with Transformers [22.995102995029576]
This paper presents an audio visual automatic speech recognition (AV-ASR) system using a Transformer-based architecture.
We focus on the scene context provided by the visual information, to ground the ASR.
Our results are comparable to state-of-the-art Listen, Attend and Spell-based architectures.
arXiv Detail & Related papers (2020-04-29T09:32:11Z)
This list is automatically generated from the titles and abstracts of the papers in this site.
This site does not guarantee the quality of this site (including all information) and is not responsible for any consequences.