Covo-Audio Technical Report
Authors: name: Wenfu Wang, name: Chenxing Li, name: Liqiang Zhang, name: Yiyang Zhao, name: Yuxiang Zou, name: Hanzhao Li, name: Mingyu Cui, name: Hao Zhang, name: Kun Wei, name: Le Xu, name: Zikang Huang, name: Jiajun Xu, name: Jiliang Hu, name: Xiang He, name: Zeyu Xie, name: Jiawen Kang, name: Youjun Chen, name: Meng Yu, name: Dong Yu, name: Rilin Chen, name: Linlin Di, name: Shulin Feng, name: Na Hu, name: Yang Liu, name: Bang Wang, name: Shan Yang
Organization: Tencent

In this work, we present Covo-Audio, a 7B-parameter end-to-end LALM that directly processes continuous audio inputs and generates audio outputs within a single unified architecture. Through large-scale curated pretraining and targeted post-training, Covo-Audio achieves state-of-the-art or competitive performance among models of comparable scale across a broad spectrum of tasks, including speech-text modeling, spoken dialogue, speech understanding, audio understanding, and full-duplex voice interaction. Extensive evaluations demonstrate that the pretrained foundation model exhibits strong speech-text comprehension and semantic reasoning capabilities on multiple benchmarks, outperforming representative open-source models of comparable scale. Furthermore, Covo-Audio-Chat, the dialogue-oriented variant, demonstrates strong spoken conversational abilities, including understanding, contextual reasoning, instruction following, and generating contextually appropriate and empathetic responses, validating its applicability to real-world conversational assistant scenarios. Covo-Audio-Chat-FD, the evolved full-duplex model, achieves substantially superior performance on both spoken dialogue capabilities and full-duplex interaction behaviors, demonstrating its competence in practical robustness. To mitigate the high cost of deploying end-to-end LALMs for natural conversational systems, we propose an intelligence-speaker decoupling strategy that separates dialogue intelligence from voice rendering, enabling flexible voice customization with minimal text-to-speech (TTS) data while preserving dialogue performance. Overall, our results highlight the strong potential of 7B-scale models to integrate sophisticated audio intelligence with high-level semantic reasoning, and suggest a scalable path toward more capable and versatile LALMs.