Frame-Wise Streaming for Efficient Learned Speech Compression
Organizations: School of Information Science and Electronic Engineering, Shanghai Jiao Tong University
Abstract
Speech compression is increasingly important in scenarios such as real-time and satellite communication. Neural codecs have shown superior Rate-Distortion (RD) performance over traditional signal-processing codecs. Models with long-range dependency encoders/decoders and entropy models outperform convolution-based structures, but they still lack streaming support since they rely on sequence-level correlations. In this paper, we propose StreamLSC, a frame-level streaming learned speech compression model built on three fundamental components of streaming networks, i.e. Boundary-Cached Causal Convolution, Intra-Frame Layer Normalization, and RNN-mode RWKV. Our model incorporates a streamable entropy quantizer module, ensuring efficient context model while preserving causal constraints. Experiments show that the streaming model achieves performance comparable to its offline counterpart, attaining state-of-the-art rate-distortion results, significant gains in subjective evaluations, -51.37% (ViSQOL) and -53.99% (PESQ) BD-RATE savings on average, and improvements of 0.2154 and 0.5156 in BD-ViSQOL and BD-PESQ.
Figures & tables
| Frame length | 80 ms | 160 ms | 320 ms | 640 ms |
|---|---|---|---|---|
| ViSQOL | 4.8257 | 4.8283 | 4.8315 | 4.8319 |
| Training Strategy | Bitrate (bps) | ViSQOL | PESQ |
|---|---|---|---|
| Full GAN Training | 3529.00 | 4.4500 | 3.1800 |
| Two-stage GAN Training | 3377.29 | 4.5398 | 3.1780 |
| Model | RTF |
|---|---|
| StreamLSC (Ours) | |
| FunCodec | |
| EnCodec | |
| DAC | |
| BigCodec |
Neural Audio Codecs
Neural audio codecs learn compact representations for compressing and reconstructing speech, music, and other sounds.
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