ai.hackcv
论文精选 65arXiv

Audio-Native Speech Recognition with a Frozen Discrete-Diffusion Language Model· 音频原生语音识别

Automatic speech recognition is dominated by autoregressive decoders that emit one token at a time. We ask whether a discrete diffusion language model can transcribe speech instead, refining a whole transcript in parallel over a small number of denoising steps. We train an audio-native interface for DiffusionGemma, a 26B mixture-of-experts model that generates text by uniform, random-token discrete diffusion rather than the absorbing-mask scheme common to recent diffusion language models. A frozen Whisper encoder supplies acoustic features, a lightweight projector maps them into the model embedding space, and low-rank adapters let the frozen backbone attend to the new modality. About 42M parameters are trained, which is 0.16 percent of the backbone. We find that the natural training objectives fail to ground the audio because their gradient reaches the projector only through attention that has already dismissed it. A connectionist temporal classification loss applied through the frozen output head breaks this deadlock. The resulting model reaches 6.6 percent word error rate on LibriSpeech test-clean, transcribes in roughly eight parallel steps regardless of utterance length, and uses a single adapter trained on six languages, which we evaluate here on English, Hindi, and Mandarin.

AI 解读论文

使用冻结的离散扩散语言模型进行音频原生语音识别。

核心方法
训练一个音频原生接口连接冻结的26B参数混合专家模型DiffusionGemma,通过轻量级投影器和低秩适配器实现音频特征到模型嵌入空间的映射,并使用连接时间分类损失来改善训练目标。
适合谁读
研究者、工程师
要解决的问题
当前自动语音识别主要依赖于自回归解码器,逐个生成词汇,效率较低。
关键实验
模型在LibriSpeech test-clean数据集上达到了6.6%的词错误率,能够约8步并行生成转录文本,且使用单个适配器训练了六种语言。
主要贡献
提出了一种新的语音识别方法,能够在少量去噪步骤中并行生成完整的转录文本,同时支持多语言适配。
意义与局限
该研究提供了一种高效且多功能的语音识别方案,但目前仅在特定数据集和语言上进行了验证,可能仍有改进空间。
领域:cs.AI作者:Harsha Vardhan Khurdula、Abhinav Kumar Singh、Yoeven D Khemlani
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