LeVo 2: Hybrid LLM-Diffusion framework for stable full-length song generation with hierarchical modeling
LeVo 2 is a new hybrid LLM-Diffusion system for controllable full-length song generation that addresses the coherence-vs-acoustics trade-off through hierarchical token prediction: a language model handles semantic planning via mixed tokens, then predicts vocal and accompaniment tracks in parallel, while a diffusion-based codec reconstructs waveforms. A key contribution is an aesthetics-guided progressive post-training schedule combining SFT, offline DPO, and semi-online DPO to separately optimize quality, controllability, and musicality. Expert listening tests show LeVo 2 outperforms open-source baselines across six subjective dimensions and approaches leading commercial systems on several metrics.
Related guides (3)
Related events (8)
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.
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.
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.
OLIVE: Joint masked latent prediction and waveform reconstruction for self-supervised speech representation learning
OLIVE (Online Latent prediction with Invariant Views and rEconstruction) is a new self-supervised speech representation learning framework that combines view-augmented masked latent prediction with waveform reconstruction under a unified objective. The reconstruction objective constrains early encoder features to retain signal-level information, while masked latent prediction shapes later contextual representations toward invariance. The authors report improvements on generation and speaker tasks, competitive performance on recognition and semantic tasks, and better waveform reconstruction compared to prior SSL approaches.
BayLing-Duplex: Native full-duplex speech dialogue using a single autoregressive LLM
Researchers introduce BayLing-Duplex, a speech language model that achieves native full-duplex interaction — simultaneous listening and speaking — using a single autoregressive LLM with no auxiliary VAD or turn-taking module. Built by fine-tuning GLM-4-Voice on 400K samples plus a lightweight DPO stage, it reaches 92% turn-taking success and 100% interruption success on InstructS2S-Eval, and improves speech-response quality substantially over Moshi. The approach adds only special tokens to the standard vocabulary, making it portable across LLM architectures without architectural changes.
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.
LESS: Adaptive mutual-stability sampling cuts diffusion LLM decoding steps by 72%
Researchers introduce LESS, a training-free adaptive sampler for diffusion large language models that treats token commitment as an online stopping problem. The method uses a joint stability rule combining confidence, persistence, and distributional stability to decide when to unmask tokens, avoiding wasted computation on already-stable positions. Evaluated on Dream-7B, LLaDA-8B, and LLaDA-1.5-8B across seven benchmarks, LESS reduces reverse denoising steps by 72.1% versus fixed-budget decoding while improving accuracy over prior adaptive samplers. The step reductions translate directly to fewer Transformer forward passes and lower wall-clock latency.
d-OPSD: First on-policy self-distillation framework tailored for diffusion LLMs
Researchers introduce d-OPSD, the first on-policy self-distillation (OPSD) framework designed specifically for diffusion large language models (dLLMs). The method addresses a fundamental mismatch between existing autoregressive OPSD approaches and dLLMs' arbitrary-order generation by using suffix conditioning on self-generated answers and step-level rather than token-level divergence supervision. Across four reasoning benchmarks, d-OPSD outperforms RLVR and SFT baselines while requiring only ~10% of the optimization steps of RLVR, suggesting strong sample efficiency gains for dLLM post-training.