Large Speech Model Enabled Semantic Communication

TL;DR

This paper introduces LargeSC, a semantic communication system leveraging large speech models for adaptive, robust, and efficient speech transmission over lossy channels, outperforming traditional methods in quality and latency.

Contribution

It proposes a novel Large Speech Model enabled Semantic Communication system with adaptive control, in-band UEP, and generative recovery, advancing the robustness and efficiency of speech transmission.

Findings

Supports bandwidths from 550 bps to 2.06 kbps.

Outperforms conventional baselines in speech quality under high packet loss.

Achieves approximately 460 ms latency for real-time deployment.

Abstract

Existing speech semantic communication systems mainly based on Joint Source-Channel Coding (JSCC) architectures have demonstrated impressive performance, but their effectiveness remains limited by model structures specifically designed for particular tasks and datasets. Recent advances indicate that generative large models pre-trained on massive datasets, can achieve outstanding performance arexhibit exceptional performance across diverse downstream tasks with minimal fine-tuning. To exploit the rich semantic knowledge embedded in large models and enable adaptive transmission over lossy channels, we propose a Large Speech Model enabled Semantic Communication (LargeSC) system. Simultaneously achieving adaptive compression and robust transmission over lossy channels remains challenging, requiring trade-offs among compression efficiency, speech quality, and latency. In this work, we employ the Mimi as a speech codec, converting speech into discrete tokens compatible with existing network architectures. We propose an adaptive controller module that enables adaptive transmission and in-band Unequal Error Protection (UEP), dynamically adjusting to both speech content and packet loss probability under bandwidth constraints. Additionally, we employ Low-Rank Adaptation (LoRA) to finetune the Moshi foundation model for generative recovery of lost speech tokens. Simulation results show that the proposed system supports bandwidths ranging from 550 bps to 2.06 kbps, outperforms conventional baselines in speech quality under high packet loss rates and achieves an end-to-end latency of approximately 460 ms, thereby demonstrating its potential for real-time deployment.