Efficient Speech Codec with Cross-Scale Residual Vector Quantized Transformers

Core Concepts

ESC, a lightweight and parameter-efficient neural speech codec, achieves high audio quality through the integration of cross-scale residual vector quantization and efficient Swin Transformer blocks, outperforming existing state-of-the-art codecs in both reconstruction quality and computational complexity.

Abstract

The paper proposes Efficient Speech Codec (ESC), a novel neural audio compression framework that leverages cross-scale residual vector quantization (CS-RVQ) and mirrored hierarchical Swin Transformer blocks. Key highlights: ESC adopts a CS-RVQ approach, which quantizes the residuals between encoder and decoder features at multiple scales, enabling coarse-to-fine decoding and improved codebook utilization. The model replaces conventional convolutional layers with efficient Swin Transformer blocks, which can effectively capture local and global audio dependencies. To address the challenge of codebook collapse in vector quantized networks, the authors introduce a pre-training stage to stabilize codebook learning. Extensive experiments demonstrate that ESC achieves comparable reconstruction quality to the state-of-the-art Descript's audio codec (DAC) while being significantly more computationally efficient, with a 9x smaller model size and 2x faster encoding speed on CPUs. The ablation study confirms the effectiveness of the proposed pre-training approach in enhancing codebook utilization and audio quality. Overall, the ESC codec presents a promising alternative to existing neural audio compression methods, striking a better balance between reconstruction fidelity and computational efficiency.

Stats

ESC has a total of 8.4M parameters. The encoding time for a 10-second, 16kHz speech signal at 9kbps is 0.78 seconds on CPU and 0.10 seconds on GPU. The decoding time for a 10-second, 16kHz speech signal at 9kbps is 0.59 seconds on CPU and 0.06 seconds on GPU.

Quotes

"ESC attains double the compression ratio of the original TFNet-CSVQ described in [18], while maintaining comparable reconstruction quality to DAC, which is currently recognized as the state-of-the-art in high-fidelity audio codecs." "Extensive results show that ESC can achieve high audio quality with much lower complexity, which is a prospective alternative in place of existing codecs."

Key Insights Distilled From

ESC: Efficient Speech Coding with Cross-Scale Residual Vector Quantized Transformers

by Yuzhe Gu,Enm... at arxiv.org 05-01-2024

https://arxiv.org/pdf/2404.19441.pdf

Deeper Inquiries

How can the proposed CS-RVQ approach be extended to handle more diverse audio types beyond speech, such as music and environmental sounds

The proposed CS-RVQ approach can be extended to handle more diverse audio types beyond speech by adapting the codec's architecture and training process to accommodate the unique characteristics of different audio signals. For music, which often contains more complex and varied patterns, the codebook design and quantization process can be optimized to capture the nuances and intricacies of musical notes and instruments. This may involve increasing the number of codebooks or adjusting the vector quantization process to better represent the frequency and timbre variations present in music. For environmental sounds, which can have a wide range of frequencies and textures, the CS-RVQ approach can be enhanced by incorporating additional features or preprocessing steps to extract relevant information from the audio signals. This could include incorporating spectrogram analysis techniques or leveraging domain-specific knowledge to improve the representation and quantization of environmental sounds. By tailoring the CS-RVQ approach to the specific characteristics of different audio types, such as music and environmental sounds, the codec can achieve higher compression efficiency and audio quality across a broader range of audio signals.

What are the potential limitations of the Swin Transformer architecture in capturing long-range temporal dependencies in audio signals, and how could this be addressed

The Swin Transformer architecture, while effective in capturing local and global dependencies in images, may face challenges in capturing long-range temporal dependencies in audio signals due to the sequential nature of audio data. Audio signals often contain temporal patterns that span across longer timeframes, requiring the model to effectively capture and process these dependencies for accurate reconstruction. One potential limitation of the Swin Transformer architecture in handling long-range temporal dependencies is the fixed-size window mechanism, which may struggle to capture the entirety of long temporal sequences efficiently. To address this limitation, the architecture could be enhanced by incorporating mechanisms such as hierarchical attention or self-attention mechanisms with adaptive window sizes. These modifications would allow the model to focus on relevant temporal contexts at different scales, enabling better capture of long-range dependencies in audio signals. Additionally, techniques like dilated convolutions or recurrent neural networks (RNNs) could be integrated into the Swin Transformer architecture to provide additional context and memory for processing long-range temporal dependencies in audio signals. By combining the strengths of different architectures, the Swin Transformer can be optimized to effectively handle the complexities of audio data and improve its performance in capturing long-range temporal dependencies.

Given the focus on computational efficiency, how could the ESC codec be further optimized for deployment on resource-constrained edge devices

To further optimize the ESC codec for deployment on resource-constrained edge devices, several strategies can be implemented to enhance its computational efficiency without compromising audio quality. One approach is to explore model quantization techniques, such as weight quantization and activation quantization, to reduce the model size and computational complexity. By quantizing the model parameters and activations to lower bit precision, the ESC codec can achieve faster inference speeds and reduced memory footprint, making it more suitable for edge deployment. Another optimization strategy is to leverage hardware acceleration technologies, such as GPU optimization or specialized hardware accelerators like Tensor Processing Units (TPUs) or Field-Programmable Gate Arrays (FPGAs). By optimizing the ESC codec to leverage the parallel processing capabilities of these hardware accelerators, the model can achieve significant speedups in inference time and overall performance on edge devices. Furthermore, techniques like model pruning and knowledge distillation can be employed to reduce the model's complexity and size while maintaining audio quality. By removing redundant parameters and distilling the knowledge from the original model into a smaller, more efficient version, the ESC codec can be optimized for deployment on resource-constrained edge devices without sacrificing performance. By implementing these optimization strategies tailored to the constraints of edge devices, the ESC codec can be further enhanced for efficient deployment in real-world applications where computational resources are limited.