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5arXiv cs.CL (Computation and Language)·15d ago

USAD 2.0: Universal audio encoder scales to 1B parameters via representation distillation

USAD 2.0 is a new universal audio encoder that integrates knowledge from both self-supervised and supervised foundation models through domain-aware distillation, extending coverage to speech, music, and general audio domains. The model scales to one billion parameters via depth scaling and adds a second-stage supervised distillation step for downstream alignment with audio LLMs. Experiments report strong or state-of-the-art results across probing and LLM-based evaluations, addressing limitations of prior multi-domain encoders like USAD and SPEAR.

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Related events (8)

6arXiv · cs.CL·19d ago·source ↗

UniAudio-Token: Semantic Speech Tokenizer with General Audio Perception for Audio-LLMs

UniAudio-Token is a framework from Tencent that extends semantic speech tokenizers—commonly used as interfaces for Audio-LLMs—to support general audio perception without sacrificing speech quality. It introduces two mechanisms: Semantic-Acoustic Primitives (SAP) for structured supervision decomposing audio into linguistic, vocal, and auditory-scene components, and Semantic-Acoustic Equilibrium (SAE), a content-aware gating mechanism that restores fine-grained acoustic details from shallow layers. Evaluations show it outperforms all single-codebook baseline tokenizers on both understanding and generation tasks when integrated with downstream LLMs. Code, training/inference scripts, and model checkpoints are publicly released.

5arXiv · cs.CL·10d ago·source ↗

AuRA: Distilling audio understanding into LLMs via LoRA adaptation

AuRA is a new method for integrating speech understanding into LLMs by distilling audio encoding capability directly into LoRA-adapted model weights, bypassing cascaded ASR-LLM pipelines. A lightweight audio embedding layer feeds speech to both an ASR encoder (teacher) and a LoRA-adapted LLM (student), with layer-wise distillation aligning hidden states. The approach claims to outperform cascaded systems, bridge-based adaptation baselines, and large-scale multimodal models on multiple speech-language benchmarks while enabling parallel end-to-end inference without large-scale multimodal training.

4arXiv · cs.CL·8d ago·source ↗

Audio-LLM-based data filtering for speech-to-speech translation via Rank-to-Distill

A new arXiv paper proposes using audio large language models to filter noisy training data for end-to-end speech-to-speech translation (S2ST). The authors introduce a two-stage Rank-to-Distill strategy: a lightweight ranker generates pseudo-labels from noisy speech pairs, which then supervise an audio-LLM to make keep/drop decisions directly from raw audio. Experiments on CVSS-C and SpeechMatrix benchmarks show up to +1.4 ASR-BLEU improvement over unfiltered baselines.

4arXiv · cs.AI·5d ago·source ↗

AudioDER: Deduplication-enhanced reasoning dataset for post-training large audio-language models

Researchers introduce AudioDER, a ~191k-sample post-training dataset for Large Audio-Language Models (LALMs) built via an acoustic similarity-based deduplication pipeline to reduce redundancy and improve corpus diversity. Each sample pairs an audio clip with a multiple-choice question, answer candidates, a caption, and a chain-of-thought rationale generated by Qwen3-30B. Post-training Qwen2-Audio-7B-Instruct on AudioDER yields consistent gains on audio reasoning benchmarks including MMAU-mini, MMSU, and MMAR. The work addresses a data quality gap in audio-language training rather than proposing a new model architecture.

5arXiv · cs.CL·11d ago·source ↗

ADAS: Attention-Discounted Adaptive Sampler improves parallel decoding for masked diffusion language models

Researchers propose ADAS, a training-free reranking rule for masked diffusion language model decoding that addresses token interaction failures in parallel token commitment. The method greedily penalizes candidates that attend strongly to already-selected uncertain positions, using attention weights as soft marginal penalties rather than hard constraints. Evaluated on LLaDA-8B-Base and Dream-7B-Base across GSM8K, MATH500, HumanEval, and MBPP, ADAS improves low-NFE performance by 9–10 percentage points on average when plugged into existing samplers with only 3.1% runtime overhead.

6arXiv · cs.AI·16d ago·source ↗

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.

5Hugging Face Blog·1mo ago·source ↗

Universal Assisted Generation: Faster Decoding with Any Assistant Model

Hugging Face introduces Universal Assisted Generation (UAG), a technique that extends speculative decoding to work with any assistant model regardless of tokenizer or vocabulary differences. The approach enables using smaller, mismatched draft models to accelerate inference of larger target models, removing the previous constraint that both models share the same tokenizer. This broadens the practical applicability of speculative decoding across the open-weights ecosystem.

4Hugging Face Blog·1mo ago·source ↗

AudioLDM 2, but faster ⚡️

Hugging Face published a blog post on AudioLDM 2, a latent diffusion model for audio generation, with a focus on inference speed improvements. The post likely covers integration into the Diffusers library and optimization techniques for faster audio synthesis. AudioLDM 2 supports text-to-audio, text-to-music, and text-to-speech generation tasks.