Jun 2026· arXiv.org· Vol abs/2606.30700· 0 citations· 28 references
Computer ScienceEngineering
TL;DR
BEST-RQ-2 is presented, an evolution of BEST-RQ that retains frozen randomprojection-based discrete targets while introducing a two-step contextualize-then-predict pretraining scheme, and consistently outperforms one-stage baselines in overall transfer while keeping inference compute unchanged.
Abstract
Self-supervised learning enables audio representations that transfer across domains and tasks. We present BEST-RQ-2, an evolution of BEST-RQ that retains frozen randomprojection-based discrete targets while introducing a two-step contextualize-then-predict pretraining scheme. A ViT context encoder processes only the unmasked spectrogram regions, and a lightweight predictor infers targets for the masked regions; the predictor is discarded after pretraining. Replacing the original Conformer encoder with a ViT shifts performance across domains, slightly reducing speech performance while improving music and environmental sounds, with comparable average scores. The main improvement comes from decomposing masked prediction into separate contextualization and prediction stages. On the X-ARES and XARES-LLM benchmarks, BEST-RQ-2 consistently outperforms one-stage baselines in overall transfer while keeping inference compute unchanged. Code and model checkpoints are publicly available.
Text-to-speech systems are improving fast, but measuring how natural they sound still requires expensive human listening tests. Existing automatic methods struggle to generalise well across different datasets. We present QMOS, a MOS prediction framework that extracts hierarchical speech quality features from a frozen Qwen2-Audio large audio language model and combines them with layer-weighted WavLM SSL representations through a learned cross-attention fusion. This design captures both high-level semantic naturalness and lowlevel acoustic distortions in a unified model. On two standard benchmarks, SOMOS and BVCC, QMOS achieves competitive system-level SRCC of 0.923 and 0.921 on SOMOS and BVCC, respectively, while using no system-ID conditioning or listener embeddings. Cross-domain evaluation yields a competitive SRCC of 0.702, suggesting the learned representations generalise across acoustic domains.
Sri Ravi Sastry Kolluru, Charan Devarakonda, S. Radhe et al.· International Conference on...· 0 citations
Self-supervised learning (SSL) has driven substantial progress in audio representation learning, though existing methods have increasingly relied on elaborate pre-training recipes to reach competitive performance. A markedly different pre-training philosophy underpins the most influential progress in language modeling and, more recently, in visual representation learning: rather than train encoders as static feature extractors, models are trained to predict the next element, a discrete token or a continuous embedding, from the preceding context. Autoregressive prediction thereby provides a unified pre-training interface that transfers across modalities, compelling the model to learn the underlying data distribution. We ask whether such a simple causal paradigm can yield strong audio learners, given that audio's temporal structure makes autoregressive prediction of patch embeddings a natural fit. We introduce NAPE (Next-Audio-Patch-Embedding prediction), a self-supervised framework in which a causal Transformer predicts each next patch embedding of a log-mel spectrogram from the previous ones, using causal masking and stop-gradient as its sole training signal. The design is intentionally minimalist, avoiding reconstruction decoders, acoustic tokenizers, student-teacher setups, and auxiliary regularization losses. Across six audio and speech benchmarks, NAPE achieves state-of-the-art fine-tuning performance on several tasks, scales consistently across encoder sizes, and yields strong linear-probing results. NAPE also produces structured attention patterns without explicit supervision.
Umberto Cappellazzo, Xubo Liu, Stavros Petridis et al.· 0 citations
Recent audio large language models (ALLMs) are typically built upon audio encoders trained with large amounts of supervised data. Since self-supervised learning (SSL) audio encoder models are known to learn general-purpose and transferable representations, we investigate whether general-purpose SSL audio representations can serve as an effective foundation for ALLMs. We present SALMONN-2, an ALLM built upon a unified SSL encoder. To better exploit the hierarchical representations learned by SSL encoders, we propose a multi-layer feature fusion (MLF) adapter that aggregates information from all encoder layers before projecting them into the language model. Beyond conventional audio understanding tasks, we further explore multimodal in-context learning (MICL) in ALLMs and study how this capability can be acquired through contextual biasing training. Experimental results show that a general-purpose SSL encoder achieves performance comparable to, or better than, specialised supervised audio encoders while providing a more balanced capability across speech, audio, music and paralinguistic tasks. SALMONN-2 further achieves state-of-the-art performance among comparable-scale open-weight models on ALLM understanding benchmarks, obtaining the best results on MMAU-Pro, MMAR and MMSU. We also show that MICL does not emerge naturally in ALLMs, but can be effectively acquired through targeted contextual biasing training.
Xiaoyu Yang, Xuenan Xu, Wenyi Yu et al.· 0 citations
It is suggested that many reported downstream gains reflect instance and seed dependent elicitation match, rather than universally improving the attainable performance ceiling, in self-supervised fine-tuning.
Wangjin Zhou, Yizhou Zhang, Yichi Wang et al.· 0 citations
The results show that a small, carefully chosen activation set can switch on test-time scaling inside a frozen ASR model without corrupting the model itself, converting fixed per-utterance compute into input-dependent compute where it is most needed.
Ho-Lam Chung, Yiming Chen, Dau-Cheng Lyu et al.· 0 citations
Automatic speech recognition (ASR) systems based on Conformer architectures achieve strong performance by jointly modeling local acoustic patterns and global contextual dependencies. However, their interleaved convolution–attention design leads to progressive entanglement of fine-grained acoustic features and global semantic representations, which weakens monotonic alignment in speech recognition and degrades performance in long utterances. To address this limitation, we propose an AS-Split Conformer–Mamba framework that decouples local and global modeling into two explicit stages. First, a stage-wise encoder is introduced, where a dedicated local modeling stage extracts phonetic-level acoustic features using SE-enhanced convolution, followed by a global modeling stage that captures long-range dependencies via multi-head self-attention and temporal convolution. Second, a Transition Fusion Block (TFB) is designed as an adaptive transition module that transforms local acoustic representations before they enter the global modeling stage. Third, intermediate CTC supervision is introduced to explicitly strengthen monotonic alignment at shallow representations. Finally, a hybrid Transformer–Mamba decoder is adopted, in which the Mamba block provides O(N) state-space computation within the replaced FFN sublayer while retaining Transformer attention mechanisms for acoustic–text alignment. Experiments conducted on AISHELL-1, THCHS-30, and ST-CMDS demonstrate that the proposed method achieves consistent improvements over strong baselines. On AISHELL-1, our model reduces Character Error Rate (CER) from 5.7% to 4.8% and Sentence Error Rate (SER) from 24.8% to 20.5%, while maintaining competitive computational efficiency.
Lulu Qin, Xuan Fu, Mingchen Sun et al.· Electronics· 0 citations