Papers for

audio transcription services

Papers whose findings have a practical use for this group, as judged from the abstract. Open a paper to read what it means in practice.

Speaker extraction model improves separation using multiple prompts

PromptSep-Speaker: Generative Target Speaker Extraction via Multimodal Prompting and Semantic Alignment

Abstract: Target speaker extraction (TSE) isolates a desired speaker from a multi-speaker mixture using cues that identify the target. Prior work has explored acoustic and textual cues, such as enrollment audio and different types of speaker descriptors, but typically relies on a single cue family. Yet no single cue is both broadly available and reliably discriminative. Moreover, most TSE systems adopt mask-prediction models, which are prone to audible artifacts and generalize poorly across acoustic conditions. To address these limitations, we propose PromptSep-Speaker, a generative TSE model based on latent diffusion that unifies four interchangeable cues: speaker temporal order, transcript, enrollment audio, and speaker caption. The generative formulation delivers higher separation quality and multi-modal prompting improves applicability across scenarios. To fully exploit each cue, we further introduce a frozen WavLM representation with an auxiliary alignment loss that supervises query-mixture cross-attention without extra labeling. Across four evaluation sets and a listening study, PromptSep-Speaker achieves state-of-the-art performance over TSE and universal source separation baselines, improving separation accuracy and quality across cue modalities.

Fri 16 OctSound
The gist
Separating one person's voice from a noisy recording with many people talking is hard. The authors created a new system that listens better by using different clues, like a person's voice sample, their speech transcript, or descriptions about them. Their model generates separated speech more clearly than older methods. It also works well in different noisy situations by combining these clues smartly.
Open → 2610.23066v1

Speech event detector identifies non-verbal sounds with timestamps

Speech event detector: Timestamped detection of non-verbal vocalizations and background sounds for rich speech transcription

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.

Fri 9 OctSound
The gist
Automatic speech recognition usually turns spoken words into text but misses sounds like laughs, coughs, or background noises. The authors present a model that can detect many different non-verbal sounds and background noises, tagging them with exact times in recordings. This model is trained on various datasets with different labeling formats and combines those labels to improve detection. It runs much faster than real time and can help make speech transcripts richer and more informative.
Open → 2610.14232v1

AnchorPrompt improves audio language model accuracy under noise and distortion

AnchorPrompt: Self-Distilled Soft Prompts for Robust Audio-Language Models

Abstract: Large audio-language models (LALMs) are sensitive to input perturbations, such as noise, waveform corruption, and adversarial injections. We propose AnchorPrompt, an efficient adaptation method that keeps the model frozen and learns a single block of prompt vectors inserted at the decoder input, between the audio and question embeddings. We train these vectors through self-distillation over diverse audio and text perturbations. To improve answer consistency and mitigate hallucination, we use the model's prediction on the clean recording as the target for answerable inputs, and assign a refusal target when the audio lacks sufficient evidence to answer. Furthermore, AnchorPrompt is perturbation-agnostic at inference, requiring no prior detection of perturbations and enabling zero-shot transfer to unseen distortions. We evaluate three LALMs across three benchmarks and show that AnchorPrompt improves answer consistency in most tested conditions. Clean accuracy improves in six of nine model-benchmark pairs, with minimal impact on the remainder of 1.2% at most. Crucially, AnchorPrompt reduces hallucinations under severe audio corruption while keeping false refusals on clean audio rare. Finally, these consistency gains transfer to unseen perturbations, such as choice permutations and reverberation.

Wed 30 SeptSoundMachine Learning
The gist
Audio-language AI systems can struggle to understand sound recordings that are noisy or distorted. The authors created AnchorPrompt, a method that teaches these AI models to stay consistent and avoid making things up when faced with unclear or corrupted audio. Instead of changing the whole model, they add a special prompt and train it to match the model’s answers on clean audio or to refuse answers when unsure. This helps the models handle new kinds of sound problems without extra preparation, improving reliability across tests.
Open → 2610.00706v1