Low Frame-rate Speech Codec: a Codec Designed for Fast High-quality Speech LLM Training and Inference

Edresson Casanova, Ryan Langman, Paarth Neekhara, Shehzeen Hussain, Jason Li, Subhankar Ghosh, Ante Jukić, Sang-gil Lee
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Abstract

Large language models (LLMs) have significantly advanced audio processing through audio codecs that convert audio into discrete tokens, enabling the application of language modeling techniques to audio data. However, audio codecs often operate at high frame rates, resulting in slow training and inference, especially for autoregressive models. To address this challenge, we present the Low Frame-rate Speech Codec (LFSC): a neural audio codec that leverages finite scalar quantization and adversarial training with large speech language models to achieve high-quality audio compression with a 1.89 kbps bitrate and 21.5 frames per second. We demonstrate that our novel codec can make the inference of LLM-based text-to-speech models around three times faster while improving intelligibility and producing quality comparable to previous models.
低帧率语音编解码器:专为快速高质量语音 LLM 训练和推理而设计的编解码器
大语言模型(LLM)通过音频编解码器将音频转换为离散的词块,大大推进了音频处理,从而使语言建模技术能够应用于音频数据。然而,音频编解码器通常以高帧率运行,导致训练和推理速度缓慢,尤其是自回归模型。为了应对这一挑战,我们提出了低帧率语音编解码器(LFSC):一种神经音频编解码器,它利用有限标量量化和对抗训练以及大型语音语言模型,以 1.89 kbps 的比特率和每秒 21.5 帧的速度实现高质量音频压缩。我们证明,我们的新型编解码器可将基于 LLM 的文本到语音模型的推理速度提高约三倍,同时提高可懂度,并产生与以前的模型相当的质量。
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