FlowEdit: Lifelong pronunciation adaptation for flow-matching TTS via associative memory
FlowEdit is a new framework enabling lifelong pronunciation correction in frozen flow-matching text-to-speech systems without retraining model weights. Corrections are stored as token-level perturbations in text embedding space within a Modern Hopfield Network, retrieved at inference via soft attention with fuzzy morphological matching. On a curated benchmark of 312 multilingual proper nouns across 18 language families, the method reduces target-word Phoneme Error Rate by 92.7% relative to the zero-shot baseline, with each correction completing in ~15 seconds on a single GPU.
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DirectAudioEdit: Training-free, inversion-free text-guided audio editing via diffusion prediction contrast
Researchers introduce DirectAudioEdit, the first training-free and inversion-free method for text-guided audio editing using diffusion denoising dynamics. The approach constructs a source-to-target editing path without requiring DDPM inversion, reducing macro-averaged FAD and KL divergence by ~16% compared to inversion-based baselines while achieving up to 64.5% speedup. Experiments span music and event-level benchmarks across two backbone architectures.
Continual learning approach for disfluency-aware ASR with explicit disfluency tokens
A new arXiv preprint addresses the challenge of transcribing disfluent speech (hesitations, repetitions, fillers) in ASR systems, which typically omit such markers causing information loss. The authors introduce explicit disfluency tokens into a pretrained ASR model and apply continual learning to adapt across datasets with varying disfluency distributions while mitigating catastrophic forgetting. The work identifies a trade-off between disfluency marker learning and general ASR performance, and finds a consistent cross-attention head mechanism shared across continual learning methods.
Corpus-Grounded Feature Diffusion pipeline for automated IEP generation in Traditional Chinese
Researchers propose a low-resource fine-tuning pipeline called Corpus-Grounded Feature Diffusion (CGFD) to automate Individualized Education Program (IEP) drafting from Traditional Chinese parent-teacher interview transcripts. The approach fine-tunes Breeze-7B with QLoRA on 582 synthetically diffused samples and uses schema-constrained decoding at inference time, finding that Grammar-Constrained Decoding is counterproductive under Traditional Chinese token budgets. On a small formal hold-out (n=10), the system achieves BERTScore F1 of 0.779, outperforming zero-shot GPT-5.4, DeepSeek-V3.2, Gemini-3-Flash-Preview, and Llama-4-Maverick baselines while enabling fully local, air-gapped inference. The work addresses a gap in Traditional Chinese special-education NLP and demonstrates a privacy-preserving deployment pattern for sensitive document generation.
Knowledge editing via locate-then-edit transferred to masked diffusion language models, revealing multi-token failure mode
A new arXiv paper investigates whether locate-then-edit knowledge editing methods, developed for autoregressive models, transfer to masked diffusion language models (MDMs) such as LLaDA and Dream. The authors find that causal tracing identifies the same early-to-mid-layer MLP location in both paradigms, but MDMs degrade systematically on multi-token edits due to partially unmasked intermediate states that the edit was never optimized for. A correction targeting these intermediate states substantially restores multi-token editing performance. The work is the first systematic comparison of knowledge editing across autoregressive and diffusion-based language model paradigms.
Audio Interaction Model: Unified Streaming LALM with Always-On Perceive-Decide-Respond Loop
Researchers introduce the Audio Interaction Model framework and a concrete implementation called Audio-Interaction, a unified streaming Large Audio Language Model that handles both offline tasks and real-time audio interaction through a continuous perceive-decide-respond loop. The system is built on SoundFlow, a framework covering data construction, training, and asynchronous low-latency inference. The authors also release StreamAudio-2M, a 2.6M-item streaming corpus spanning 28 sub-tasks, and Proactive-Sound-Bench for evaluating proactive audio intervention. Evaluated across 8 benchmarks, the model preserves competitive offline performance while enabling real-time ASR, streaming instruction following, and proactive response capabilities not available in prior offline LALMs.
DOA: Training-Free Decoder-Only Attention Policy for Long-Form Simultaneous Speech Translation with SpeechLLMs
The paper proposes Decoder-Only Attention (DOA), a training-free streaming policy for simultaneous speech-to-text translation (SimulST) that works with off-the-shelf decoder-only Speech LLMs. DOA derives proxy alignment signals from self-attention rather than cross-attention, enabling long-form simultaneous translation without retraining. Experiments on Phi4-Multimodal and Qwen3-Omni demonstrate low-latency performance approaching offline decoding quality, validating that decoder self-attention contains sufficient alignment information for streaming decisions.
Fine-tuning Florence-2 - Microsoft's Cutting-edge Vision Language Models
This Hugging Face blog post provides a technical guide for fine-tuning Microsoft's Florence-2 vision-language models. Florence-2 is a compact yet capable multimodal model supporting tasks like captioning, object detection, and OCR. The post covers practical implementation details for adapting the model to custom datasets using the Hugging Face ecosystem.
Study finds optimal speech token frame rate for aligning speech with text-native LLM reasoning
Researchers identify a temporal-granularity mismatch as a key cause of reasoning degradation in spoken dialogue models: speech tokens are far longer than text under matched semantics, diluting per-token semantic density. The paper introduces factorized FSQ and a non-autoregressive audio LM head to enable low frame rates, then sweeps frame rates from 50Hz down to 2.08Hz under a frozen LLM backbone. Results show a consistent optimal regime at 4.17Hz with intermediate-layer representation alignment for speech QA tasks.