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A Transcription Prompt-based Efficient Audio Large Language Model for Robust Speech Recognition
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Audio-LLM introduces audio modality into a large language model (LLM) to enable a powerful LLM to recognize, understand, and generate audio. However, during speech recognition in noisy environments, we observed the presence of illusions and repetition issues in audio-LLM, leading to substitution and insertion errors. This paper proposes a transcription prompt-based audio-LLM by introducing an ASR expert as a transcription tokenizer and a hybrid Autoregressive (AR) Non-autoregressive (NAR) decoding approach to solve the above problems. Experiments on 10k-hour WenetSpeech Mandarin corpus show that our approach decreases 12.2% and 9.6% CER relatively on Test_Net and Test_Meeting evaluation sets compared with baseline. Notably, we reduce the decoding repetition rate on the evaluation set to zero, showing that the decoding repetition problem has been solved fundamentally.
Forward citations
Cited by 3 Pith papers
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SimulS2ST-Omni: Data-Efficient Streaming Speech-to-Speech Translation via Explicit Trajectory Supervision
A joint text-code trajectory supervision recipe lets a two-stream speech LM achieve competitive long-form streaming S2ST with ~2k hours of paired speech.
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MFLA: Monotonic Finite Look-ahead Attention for Streaming Speech Recognition
MFLA adds finite look-ahead attention plus a CIF-based token counter to Whisper, enabling streaming recognition with a wait-k latency-quality trade-off.
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Leveraging LLM for Stuttering Speech: A Unified Architecture Bridging Recognition and Event Detection
An LLM-driven multi-task system reports a 5.45% CER and 73.63% average SED F1 on the AS-70 Mandarin stuttering benchmark, though key baselines and uncertainty are missing.
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