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 baichuan-audio


S2SBench: A Benchmark for Quantifying Intelligence Degradation in Speech-to-Speech Large Language Models

Fang, Yuanbo, Sun, Haoze, Liu, Jun, Zhang, Tao, Zhou, Zenan, Chen, Weipeng, Xing, Xiaofen, Xu, Xiangmin

arXiv.org Artificial Intelligence

End-to-end speech large language models ((LLMs)) extend the capabilities of text-based models to directly process and generate audio tokens. However, this often leads to a decline in reasoning and generation performance compared to text input, a phenomenon referred to as intelligence degradation. To systematically evaluate this gap, we propose S2SBench, a benchmark designed to quantify performance degradation in Speech LLMs. It includes diagnostic datasets targeting sentence continuation and commonsense reasoning under audio input. We further introduce a pairwise evaluation protocol based on perplexity differences between plausible and implausible samples to measure degradation relative to text input. We apply S2SBench to analyze the training process of Baichuan-Audio, which further demonstrates the benchmark's effectiveness. All datasets and evaluation code are available at https://github.com/undobug/S2SBench.


Baichuan-Audio: A Unified Framework for End-to-End Speech Interaction

Li, Tianpeng, Liu, Jun, Zhang, Tao, Fang, Yuanbo, Pan, Da, Wang, Mingrui, Liang, Zheng, Li, Zehuan, Lin, Mingan, Dong, Guosheng, Xu, Jianhua, Sun, Haoze, Zhou, Zenan, Chen, Weipeng

arXiv.org Artificial Intelligence

We introduce Baichuan-Audio, an end-to-end audio large language model that seamlessly integrates audio understanding and generation. It features a text-guided aligned speech generation mechanism, enabling real-time speech interaction with both comprehension and generation capabilities. Baichuan-Audio leverages a pre-trained ASR model, followed by multi-codebook discretization of speech at a frame rate of 12.5 Hz. This multi-codebook setup ensures that speech tokens retain both semantic and acoustic information. To further enhance modeling, an independent audio head is employed to process audio tokens, effectively capturing their unique characteristics. To mitigate the loss of intelligence during pre-training and preserve the original capabilities of the LLM, we propose a two-stage pre-training strategy that maintains language understanding while enhancing audio modeling. Following alignment, the model excels in real-time speech-based conversation and exhibits outstanding question-answering capabilities, demonstrating its versatility and efficiency. The proposed model demonstrates superior performance in real-time spoken dialogue and exhibits strong question-answering abilities. Our code, model and training data are available at https://github.com/baichuan-inc/Baichuan-Audio