End-to-end Streaming model for Low-Latency Speech Anonymization

Waris Quamer1, Ricardo Gutierrez-Osuna1

1Department of Computer Science and Engineering, Texas A&M University, USA

Abstract

Speaker anonymization aims to conceal cues to speaker identity while preserving linguistic content. Current machine learning based approaches require substantial computational resources, hindering real-time streaming applications. To address these concerns, we propose a streaming model that achieves speaker anonymization with low latency. The system is trained in an end-to-end autoencoder fashion using a lightweight content encoder that extracts HuBERT-like information, a pretrained speaker encoder that extract speaker identity, and a variance encoder that injects pitch and energy information. These three disentangled representations are fed to a decoder that re-synthesizes the speech signal. We present evaluation results from two implementations of our system, a full model that achieves a latency of 230ms, and a lite version (0.1x in size) that further reduces latency to 66ms while maintaining state-of-the-art performance in naturalness, intelligibility, and privacy preservation.

Block Diagram

Block Diagram
Block diagram of the proposed system (a) Training flow (b) Inference flow.

Notes

Audio Samples

**Real-time Latency marked in red.

Chunk Size Latency Input speech Base Lite
20ms
(ms) base lite
CPU 87 44
GPU 45 39
40ms
(ms) base lite
CPU 118 65
GPU 65 60
80ms
(ms) base lite
CPU 193 109
GPU 105 100
160ms
(ms) base lite
CPU 284 206
GPU 187 181
240ms
(ms) base lite
CPU 373 334
GPU 267 262
320ms
(ms) base lite
CPU 476 443
GPU 350 343