https://github.com/idiap/rnv
공식 코드Unsupervised Rhythm and Voice Conversion to Improve ASR on Dysarthric Speech
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Automatic Speech Recognition (ASR) involves converting spoken language into written text. It is designed to transcribe spoken words into text in real-time, allowing people to communicate with computers, mobile devices, and other technology using their voice. The goal of Automatic Speech Recognition is to accurately transcribe speech, taking into account variations in accent, pronunciation, and speaking style, as well as background noise and other factors that can affect speech quality.
관련 논문과 평가 결과에 연결된 구현입니다. 출처에서 확인된 저자 구현에는 공식 코드 표시가 있습니다.
Unsupervised Rhythm and Voice Conversion to Improve ASR on Dysarthric Speech
End-to-End Speech Recognition With Joint Dereverberation Of Sub-Band Autoregressive Envelopes
NusaCrowd: Open Source Initiative for Indonesian NLP Resources
Improving LSTM-CTC based ASR performance in domains with limited training data
Consistent Training and Decoding For End-to-end Speech Recognition Using Lattice-free MMI
Towards Unsupervised Speech Recognition Without Pronunciation Models
Channel-Aware Domain-Adaptive Generative Adversarial Network for Robust Speech Recognition
Vibravox: A Dataset of French Speech Captured with Body-conduction Audio Sensors
A low latency attention module for streaming self-supervised speech representation learning
Cross-domain Speech Recognition with Unsupervised Character-level Distribution Matching
Confidence Score Based Speaker Adaptation of Conformer Speech Recognition Systems
K-Wav2vec 2.0: Automatic Speech Recognition based on Joint Decoding of Graphemes and Syllables
Speech collage: code-switched audio generation by collaging monolingual corpora
Distilling a Pretrained Language Model to a Multilingual ASR Model
Audio-Visual Representation Learning via Knowledge Distillation from Speech Foundation Models
Predicting Multi-Codebook Vector Quantization Indexes for Knowledge Distillation
A Parallelizable Lattice Rescoring Strategy with Neural Language Models
TED-LIUM 3: twice as much data and corpus repartition for experiments on speaker adaptation
Data augmentation using prosody and false starts to recognize non-native children's speech
A context-aware knowledge transferring strategy for CTC-based ASR
DuplexMamba: Enhancing Real-time Speech Conversations with Duplex and Streaming Capabilities
Whose Emotion Matters? Speaking Activity Localisation without Prior Knowledge
Squeezeformer: An Efficient Transformer for Automatic Speech Recognition
wav2graph: A Framework for Supervised Learning Knowledge Graph from Speech
출처: Papers with Code · CC BY-SA 4.0. 아카이브 자료와 새로 수집한 논문은 별도로 관리합니다.