Separate and Reconstruct: Asymmetric Encoder-Decoder for Speech Separation

In speech separation, time-domain approaches have successfully replaced the time-frequency domain with latent sequence feature from a learnable encoder. Conventionally, the feature is separated into speaker-specific ones at the final stage of the network. Instead, we propose a more intuitive strategy that separates features earlier by expanding the feature sequence to the number of speakers as an extra dimension. To achieve this, an asymmetric strategy is presented in which the encoder and decoder are partitioned to perform distinct processing in separation tasks. The encoder analyzes features, and the output of the encoder is split into the number of speakers to be separated. The separated sequences are then reconstructed by the weight-shared decoder, which also performs cross-speaker processing. Without relying on speaker information, the weight-shared network in the decoder directly learns to discriminate features using a separation objective. In addition, to improve performance, traditional methods have extended the sequence length, leading to the adoption of dual-path models, which handle the much longer sequence effectively by segmenting it into chunks. To address this, we introduce global and local Transformer blocks that can directly handle long sequences more efficiently without chunking and dual-path processing. The experimental results demonstrated that this asymmetric structure is effective and that the combination of proposed global and local Transformer can sufficiently replace the role of inter- and intra-chunk processing in dual-path structure. Finally, the presented model combining both of these achieved state-of-the-art performance with much less computation in various benchmark datasets.

Paper

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Peer review

Reviewer y9eH5/10 · confidence 3/52024-06-24

Summary

This paper proposes a novel transformer based architecture, sepreformer, for speech separation based on a siamese decoder network that operates on separated speech signals in the encoded space.

Strengths

A major strength of the paper is it's state of the art performance on a variety of datasets, showing as high as 25dB on the WSJ 2 mix. In addition, sepreformer is quite computationally efficient when compared with other baseline methods. The supplementary materials include wav files which show the fidelity of the reconstructed examples. The ablation studies provide good insights into the contributions of different component, such as the depth of the encoder decoder. Details of the method are provided with high degree of reproducibility and description.

Weaknesses

The first major weakness is the presentation quality of the paper. Starting with the abstract, which goes right into a discussion of feature length and computation without setting up the problem and overview. I think a lot of the high level picture is missing, especially related to the particular insights that help this method work better. Beyond the presentation, a main concern I have of this method is extending to beyond 2 sources. The experiments focus on two-speaker separation. It's unclear how well the method would generalize to scenarios with more than two speakers, which is an important real-world use case. It's also particularly important here because the authors propose separating the sources first and then using a siamese network for decoding the features back to speech. With more than 2 speakers, it's quite likely that the method would have a harder time as the features for each source get separated out earlier in the process. The experiments and comparisons primarily focuses on SI-SNRi. I'd like to see other metrics like PESQ or STOI considered as SNR does not always represent perceptual quality.

Questions

Are there any experiments conducted on more than 2 speakers? Are there any metrics considered beyond SNR?

Rating

5

Confidence

3

Soundness

3

Presentation

1

Contribution

3

Limitations

No

Reviewer UNNK6/10 · confidence 5/52024-07-03

Summary

This paper presents a novel approach to time-domain speech separation, departing from the conventional chunk-based dual-path processing. The authors introduce an asymmetric encoder-decoder architecture, where the encoder analyzes features and splits them based on the number of speakers. A Siamese decoder reconstructs the separated sequences, learning to discriminate features without explicit speaker information. The use of global and local Transformer blocks for long-sequence processing eliminates the need for chunking, contributing to a more efficient and effective model.

Strengths

1. The paper introduces an innovative asymmetric encoder-decoder framework for time-domain speech separation, deviating from the standard chunk-based dual-path models. 2. Efficient Feature Discrimination: The Siamese decoder enables the model to learn to discriminate features directly without relying on explicit speaker information, leading to a more streamlined and potentially more robust separation process. 3. The proposed model achieves good performance on benchmark datasets while requiring significantly less computation than previous approaches. This demonstrates the potential for this method to be applied in real-world applications where computational resources may be limited.

Weaknesses

The paper presents some interesting ideas, but their novelty and significance are questionable: 1. Transformer Usage: While the use of Transformer blocks is highlighted, similar architectures have been successfully employed in previous works like Sepformer, raising questions about the uniqueness of this contribution. 2. Limited Evaluation: The experimental results primarily focus on two-speaker separation, which is considered a relatively solved problem in the current state-of-the-art. The absence of evaluations on more challenging scenarios with three or more speakers limits the generalizability and impact of the findings. 3. Incomplete Comparison: The paper's claims of achieving state-of-the-art results are undermined by the lack of comparison with other important papers in the field. Notably, models like the "DIFFUSION-BASED SIGNAL REFINER FOR SPEECH SEPARATION" have reported superior performance (SI-SDR of 23.1dB), raising concerns about the validity of the SOTA claim.

Questions

--

Rating

6

Confidence

5

Soundness

3

Presentation

3

Contribution

2

Limitations

--

Reviewer aFMn6/10 · confidence 4/52024-07-12

Summary

The paper proposes SepReformer, an efficient time-domain separation network. The model is an encoder-decoder architecture that splits the output features of the encoder based on the number of speakers before feeding them to the decoder. Both encoder and decoder networks are comprised of transformer blocks that capture global and local characteristics of the signal in different time-scales. The proposed approach achieves state of the art results in 3 well-established datasets.

Strengths

1. Paper well-written and easy to follow. 2. Efficient architecture that produces state of the art results. 3. Thorough experimentation and ablation analysis of their approach.

Weaknesses

1. Testing on clean data. It would be interesting to see how the proposed model performs on noisy datasets. 2. Evaluation of the approach to the two-speaker separation problem.

Questions

1. How does the early split strategy affect the overall computational efficiency compared to late split and other conventional methods? 2. Why was the alpha value set to 0.4, and have you experimented with other values to determine the optimal setting? 3. How does the model handle varying lengths of input sequences, and what is the maximum sequence length it can effectively process? 4. Are there any limitations or failure modes of the proposed method that have been identified during experimentation?

Rating

6

Confidence

4

Soundness

3

Presentation

3

Contribution

3

Limitations

1. The paper does not discuss potential failure modes or limitations observed during experimentation,

Reviewer Gfro7/10 · confidence 5/52024-07-13

Summary

The authors propose a neural network architecture to separate a speech mixture containing 2 speakers. The proposed U-net based architecture replaces the inter and intra chunk processing - a popular method for speech separation - with global and local attention mechanisms. They further propose a mechanism to reduce the computational cost of the model. The model does the speech separation early in the network and uses a decoder with a low parameter count - all thanks to its weight sharing strategy - to do the speech separation. Evaluation on simulated data shows improvement over the state of the art methods.

Strengths

1. The Efficient Global Attention (EGA) component of the model, as discussed in Section 3.3, appears to be a cost-effective method for using the global context. This is achieved by initially subsampling to a reduced number of frames, thereby reducing the computational cost associated with transformers, and subsequently upsampling to a larger number of frames. The anticipated loss incurred from the downsampling process is compensated through the implementation of a gating mechanism. 2. Results show all the proposed mechanics, namely: early split, multi-loss, shared decoder parameters, EGA module design gives improvement in Si-SDR 3. Interestingly, when the proposed methods were applied to existing architectures such as conv-tasnet and Sepformer, an improvement in SI-SNR was observed. 4. Appendix D, also shows the computation effectiveness of the model 5. Attached samples clearly showed the quality of separated speech using the proposed network.

Weaknesses

1. The authors have shown good results on a bunch of datasets, and the separated audios in the supplemental file are of high quality. But all these results are based on simulated data, so it makes you wonder how well the model would do with real data. It’d be great if the authors could show how the model performs on the Chime-6 dataset or Libricss, maybe through WER metrics or even objective speech perception measures after separation. Or at least, they could show us some samples after separation on real data.

Questions

1. Looking at Figure 2, I’m wondering if it’s really a good idea to use the same split network for all ‘r’ values. Would it be better to have a different ‘spk’ split for each ‘r’ ? 2. I believe Eq(1) should be a minimum of the Si-SNR and \tau since you refer to it in line 201? 3. In line 213, shouldn’t the multi loss be (1-\alpha / R) L + \alpha \sum_r L_r/R ? 4. It is not very clear why the authors refer to the decoder as a Siamese decoder. Is this because they share the same weights across all speakers? This is typically how a speech separation network is structured. I don't see any contrastive learning loss functions as part of the training loss to discriminate the speakers.

Rating

7

Confidence

5

Soundness

3

Presentation

2

Contribution

3

Limitations

Lack of evaluation on real data is a limitation of this work.

Reviewer Gfro2024-08-13

Thanks for addressing the comments.

I thank the authors for providing their rebuttal which address my concerns. I will retain the Accept score.

Program Chairsdecision2024-09-25

Decision

Accept (poster)

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