Q-SPT is proposed, a low-frame-rate dual-stream speech tokenizer with separate, context-aware, learnable query-based compressors specialized for semantic and acoustic representations that yields the best speech recognition accuracy and text-to-speech perceptual quality with competitive intelligibility.
Abstract
Neural speech codecs increasingly serve as tokenizers for speech language models (SLMs). Lowering the frame rate reduces the computational and memory costs of SLMs, but makes it difficult to preserve both linguistic information and acoustic detail. Existing approaches rely on rule-based compression: average pooling can discard linguistic information, whereas similarity-based merging uses a fixed threshold on adjacent-frame similarity and applies the resulting boundaries to the acoustic stream. We propose Q-SPT, a low-frame-rate dual-stream speech tokenizer with separate, context-aware, learnable query-based compressors specialized for semantic and acoustic representations. In particular, queries at a fixed rate independently attend to the semantic and acoustic streams as separate key-value sources, enabling stream-specific, context-aware aggregation through two separately learned compressors. In addition, an autoregressive text loss explicitly supervises the semantic compressor to preserve linguistic information. Experimental results show that Q-SPT achieves the best reconstruction among the evaluated codecs at the same frame rate. In downstream SLMs, it yields the best speech recognition accuracy and text-to-speech perceptual quality with competitive intelligibility.
This work presents TontaubeV1, a model that preserves natural prosody while enabling streaming from a single consumer GPU, and is designed primarily for English and German, with additional multilingual support.
Aligned Continuous Integrate-and-Fire, a highly efficient framework for zero-shot speech processing that dynamically compresses continuous acoustic frames into the exact discrete token length of the target text utilizing explicit Dynamic Time Warping alignments is introduced.
Abderrahmane Issam, Yusuf Can Semerci, Jan Scholtes et al.· 0 citations
Audio large language models (ALLMs) can understand speech content, yet their ability to use speaker identity for verification remains limited. We propose TS-SP (Two-Stage Speaker Preservation), a parameter-efficient framework for learning speaker-preserving representations and making them accessible to an ALLM's langua...
Jun-Jie Li, Zheng Liang, Zhe Li et al.· 0 citations
This paper describes the system for Task~2 of the second Multilingual Conversational Speech Language Model (MLC-SLM) Challenge, which adapt Qwen3-Omni-30B-A3B-Instruct with a segment-evidence-aware data and post-training pipeline and obtains 90.92% accuracy on the final official evaluation set.
H. Le, L. Nguyen, Minh Tri Dao· 2nd Workshop on Multilingual...· 1 citation
Speech large language models (LLMs) are often built using pretrained acoustic encoders as an audio interface to an LLM decoder. However, LLMs scale poorly with input sequence length, while acoustic encoders tend to produce substantially longer sequences than typical textual inputs. This motivates learning to compress e...
Ci-Han Xiao, Yuan Tseng, Matthew Wiesner et al.· 0 citations
Factorized DSM is proposed, which separates the waiting probability for ASR-specific tokens from the text prediction space and allows the large-vocabulary softmax to be skipped on waiting steps and greatly reduces GPU memory use while maintaining similar training throughput.
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