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arXiv cs.CL· Richard Yucheng He, Chen Xu, Yihang Liu, Tairan Chen, Baodong Cao·· 12 小时前AI 评分40

InteractionBench:在真实语音系统中测量时序、任务准确率、耳语与音量及接地性

InteractionBench: Measuring Timing, Task Accuracy, Whisper and Volume, and Grounding in Real-Time Voice Systems

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研究者发布 InteractionBench,一个在同一批录制用户音频上同时测量对话时序、口语任务准确率、耳语与音量、以及二十轮以上对话接地性的基准与评测框架,含 13 个场景、每系统 221 项。

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Abstract:Most benchmarks for real-time voice systems focus on one aspect, such as turn-taking, instruction following, paralinguistic control, or conversational memory, each with its own data and set of systems. We present InteractionBench, a benchmark and evaluation harness that measures four dimensions on the same systems from the same recorded user audio: conversational timing, spoken-task accuracy, whisper and volume, and groundedness in conversations of more than twenty turns. The benchmark contains 221 items per system in thirteen scenarios. Before any system runs, we fix the expected floor-control actions for spoken tasks by rule and take the expected answers from scripted dialogues. We score whispering from the audio signal. We evaluate GPT-Live, GPT-Realtime, Gemini Live, Moshi, PersonaPlex, Freeze-Omni, and a turn-based cascade in 1,519 graded runs. Among the six full-duplex systems, timing and spoken-task accuracy are rank-correlated ($\rho = 0.83$), while the cascade has the highest spoken-task score (38/50) and the second-lowest timing score. No system corrects a user's grammar within the required window. Only Gemini Live whispers on most requests (8/10). Hosted systems pass 67-88% of groundedness probes, compared with 14-35% for open full-duplex models. Timing is measured where the client hears each system, so the timing of hosted systems includes network delay: for GPT-Live, whose server reports its own clock, removing a delay of about 1.5 s lowers its timing score from 86% to 56%. We also document where the protocol differs across systems, including the text instructions.
Comments: 21 pages, 1 figure, 14 tables
Subjects: Sound (cs.SD); Computation and Language (cs.CL); Audio and Speech Processing (eess.AS)
Cite as: arXiv:2611.02430 [cs.SD]
  (or arXiv:2611.02430v1 [cs.SD] for this version)
  https://doi.org/10.48550/arXiv.2611.02430

arXiv-issued DOI via DataCite (pending registration)

Submission history

From: Tairan Chen [view email]
[v1] Mon, 2 Nov 2026 02:53:57 UTC (27 KB)

来源:arXiv cs.CL · arxiv.org