Speech event detector: Timestamped detection of non-verbal vocalizations and background sounds for rich speech transcription
AuthorsAyush Kumar Bar, Promila Ghosh
A lightweight audio model detects and timestamps vocalizations and background sounds to make speech transcripts more complete.
Key results
Trainable parameters in the detector.
Twenty speaker vocalizations and 34 background sounds.
Milliseconds between output predictions.
Sounds located by forced alignment, out of 8,620.
Macro precision was 30.2%; the profile found 67% of 9,367 annotated events.
Times real time on eight CPU threads.
What the paper found
The Speech Event Detector adds timestamped non-verbal vocalizations and background sounds to word-focused transcripts from Whisper. Its 5.8M-parameter convolutional–Transformer predicts 54 event classes every 40 ms, including coughs, laughter, horns, and music. To combine seven public datasets with incompatible labels, the system maps them into one taxonomy and uses MMS CTC forced alignment to recover timestamps for 6,360 of 8,620 sounds in NonverbalTTS. Training mixes real recordings with synthetic events placed over speech and noise; at inference, per-class thresholds, hysteresis, and event joining turn frame probabilities into tags that are merged into Whisper’s word timestamps without changing word order. On held-out recordings, across 48 classes with at least 10 events, the sensitive profile achieved 49.1% macro event recall and 30.2% macro precision, finding 67% of 9,367 annotated events. The balanced profile reached 36.9% recall and 36.1% precision. The detector runs at 39× real time on eight CPU threads. Results should be treated cautiously: thresholds were tuned on held-out clips overlapping the evaluation recordings, and no standard benchmark was used.
Original abstract
Automatic speech recognition produces words but drops the non-verbal content of a recording: the breaths, coughs, sighs and laughter of the speaker, and the horns, music or barking around them. We describe Speech Event Detector, a 5.8-million-parameter convolutional-Transformer model that outputs probabilities every 40 ms for 54 event classes (20 speaker vocalizations and 34 background sounds), together with post-processing that converts these probabilities into timestamped events and a procedure for merging the events into a Whisper transcript. The model is trained on material drawn from seven public datasets whose labels come in incompatible forms: single detected regions, free-text tags with timestamps, temporally strong ontology labels, symbols embedded in transcripts, isolated recordings and environmental recordings. We unify them using rule-based label mapping, recover timestamps for transcript-embedded labels through CTC forced alignment, locating 6,360 of 8,620 sounds, and combine about 168 hours of real annotated training recordings with synthetic mixtures and augmentation using real background noise. Per-class thresholds for two operating profiles are tuned on real held-out recordings, and performance is estimated using two-fold cross-validation. On whole held-out recordings from four datasets, considering the 48 classes with at least ten held-out events, the sensitive profile detects an average of 49 percent of a class's events, covering 67 percent of all 9,367 events, at a macro precision of 0.30. The balanced profile achieves a macro recall of 0.37 at a precision of 0.36, with macro event F1 scores of 0.32 and 0.33, respectively. The detector runs about 39 times faster than real time on eight CPU threads.
Read the original paperMore in Speech AI
Browse all 33 papers →Fish Audio Drama 3 Technical Report
Dawei Han, Shidong Li, Shijia Liao, Jiahua Liu, Songting Liu, Qingzheng Wang, Yuxuan Wang, Xin Chen, Yifan Cheng, Tianyu Li, Gangin Park, Xuehao Yang, Chang Zeng, Haoyang Zhang, Ruoyi Zhang, Xiangyu Zhang
Fish Audio Drama 3 is an open-source speech model designed to perform, revise, and edit expressive multi-character audio from text.
Learning Where to Split: Dynamic Band Splitting for Audio Source Separation
Bingzhe Zhang, Kai Li, Xiaolin Hu
The method learns where to split an audio spectrum for more effective source separation without increasing the number of band tokens.
Accent Analogy Guidance: More Speaker Similarity at Equal Accent in Cross-Lingual Voice Cloning
Yoomee Cho, Jisun Lee
A training-free technique helps voice-cloning systems preserve who is speaking while making their accent sound native to the target language.