20 results for “Fundamentals of voice AI, natural language processing”
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Youngwon Choi, Hyeonyu Kim, Taeyoun Kwon, Donghyuk Jung +1 more
CORTIS is a text-only adaptation framework that fine-tunes spoken language models for task-oriented voice agents using text-form task supervision.
Jing Peng, Junhao Du, Chenghao Wang, Hanqi Li +20 more
The paper introduces SURE, a unified framework designed to standardize and improve the comparability and reproducibility of evaluations for advanced speech understanding models.
David Ayllon, Alice Baird, Jeffrey Brooks, Franc Camps-Febrer +10 more
The paper introduces the Real World Voice EQ Bench, a multidimensional benchmark for evaluating voice AI across text-to-speech, speech-to-speech, speech understanding, and automatic speech recognition…
Sicheng Yang, Shulan Ruan, Shiwei Wu, Yu Liu +3 more
PolySpeech-100 introduces a massive, multi-lingual benchmark covering 110 linguistic variants to rigorously test Speech-LLMs, demonstrating that open-source models struggle with low-resource languages…
This paper introduces Voice Memory, an inference-only scheme for agentic speech recognition that uses a frozen corrector and a score-gated optimizer to improve speech recognition performance.
The paper introduces BEA-Dialogue+, an expanded 200-hour corpus for Hungarian conversational ASR, demonstrating that while larger data is challenging, specialized fine-tuning techniques significantly…
The paper introduces SPEARBench, a benchmark for evaluating naturalness in speech-to-speech language models using a multidimensional protocol.
Seolhee Lee, Minsu Kang, Yangsun Lee, Woosun Min +2 more
The paper introduces the Designed Vocalizations Dataset for AI-based voice conversion research on non-human vocalizations and effects, providing a standardized test set and benchmark results.
This paper explores the effect of conversational timing properties on automatic speech recognition (ASR) systems by controlling and optimizing pause and overlap timing distributions.
Yuyue Wang, Xihua Wang, Xin Cheng, Yijing Chen +1 more
The paper introduces PlanAudio, a unified LLM-based framework that directly synthesizes natural, composite audio containing speech and sounds from unconstrained free-form text prompts, outperforming e…
Yujie Tu, Yifan Yang, Tianrui Wang, Yanqiao Zhu +32 more
The paper introduces GigaSpeechBench, a comprehensive multilingual and multidimensional ASR & AST benchmark with 680 hours of human-annotated speech, featuring 12 low-resource languages, 6 Chinese dia…
Sujith Pulikodan, Agneedh Basu, Pavan Kumar, Pranav Bhat +3 more
This paper investigates the effectiveness of incorporating synthetic speech data in Automatic Speech Recognition (ASR) Systems for three Indic languages by analyzing performance gains, script sources,…
Caption Studio is a transparency-first speech and audio intelligence platform that provides automated transcription, speaker diarization, speech analytics, signal-level audio analysis, and subtitle ge…
Mohammad Zeineldeen, Albert Zeyer, Haoran Zhang, Robin Schmitt +2 more
This paper investigates the relationship between language model perplexity and word error rate in modern automatic speech recognition systems, studying the impact of external language models, encoder…
The paper introduces Dialogs, a new Russian conversational speech corpus with high-quality recordings, segmented utterances, and expressive prosody labels.
Ke-Han Lu, Keqi Deng, Ruchao Fan, Rui Zhao +1 more
The paper proposes SpeechKV, a method to compress speech sequences inside large language models using a learned pooling, maintaining performance and delivering decoding speedup.
SooHwan Eom, Hee Suk Yoon, Eunseop Yoon, Mark Hasegawa-Johnson +1 more
The paper proposes RTFree-F5, a method to make flow-matching TTS models like F5-TTS independent of reference transcripts, improving performance and naturalness for dysarthric speakers.
Zixuan Jiang, Yanqiao Zhu, Peng Wang, Qinyuan Chen +7 more
The paper proposes Agentic ASR, a closed-loop framework that treats ASR as a multi-turn refinement task, significantly improving semantic accuracy over traditional token-level metrics.
A Wave-U-Net model is trained to extract a fundamental waveform from input speech signals for accurate and robust instantaneous pitch estimation.