Recently, conditional score-based diffusion models have gained significant
attention in the field of supervised speech enhancement, yielding
state-of-the-art performance. However, these methods may face challenges when
generalising to unseen conditions. To address this issue, we introduce an
alternative approach that operates in an unsupervised manner, leveraging the
generative power of diffusion models. Specifically, in a training phase, a
clean speech prior distribution is learnt in the short-time Fourier transform
(STFT) domain using score-based diffusion models, allowing it to
unconditionally generate clean speech from Gaussian noise. Then, we develop a
posterior sampling methodology for speech enhancement by combining the learnt
clean speech prior with a noise model for speech signal inference. The noise
parameters are simultaneously learnt along with clean speech estimation through
an iterative expectationmaximisation (EM) approach. To the best of our
knowledge, this is the first work exploring diffusion-based generative models
for unsupervised speech enhancement, demonstrating promising results compared
to a recent variational auto-encoder (VAE)-based unsupervised approach and a
state-of-the-art diffusion-based supervised method. It thus opens a new
direction for future research in unsupervised speech enhancement.

最近，条件评分驱动扩散模型在监督式语音增强领域引起了显著关注，取得了最先进的性能。然而，这些方法在泛化到未见条件时可能面临挑战。为了解决这个问题，我们引入了一种在无监督方式下操作、利用扩散模型的生成能力的替代方法。具体而言，训练阶段，在短时傅里叶变换（STFT）领域中使用评分驱动扩散模型学习了干净语音的先验分布，使其能够从高斯噪声中无条件地生成干净语音。然后，我们开发了一种通过将学习的干净语音先验与噪声模型结合起来进行语音增强的后验采样方法。噪声参数通过迭代的期望最大化（EM）方法与干净语音估计同时学习。据我们所知，这是第一个探索基于扩散的生成模型用于无监督语音增强的工作，与最近的变分自编码器（VAE）无监督方法和最先进的基于扩散的监督方法相比，取得了有希望的结果。因此，它为未来的无监督语音增强研究开辟了新的方向。

基于扩散型生成模型的无监督语音增强

Unsupervised speech enhancement with diffusion-based generative models

Speech derverberation using a single microphone is addressed in this paper.
Motivated by the recent success of the fully convolutional networks (FCN) in
many image processing applications, we investigate their applicability to
enhance the speech signal represented by short-time Fourier transform (STFT)
images. We present two variations: a "U-Net" which is an encoder-decoder
network with skip connections and a generative adversarial network (GAN) with
U-Net as generator, which yields a more intuitive cost function for training.
To evaluate our method we used the data from the REVERB challenge, and compared
our results to other methods under the same conditions. We have found that our
method outperforms the competing methods in most cases.

本文利用全卷积网络 (FCN) 来处理语音信号的混响问题，提出了用 U-Net 和 GAN 算法来处理时频短时傅里叶变换 (STFT) 表示的语音信号。通过与其他算法的比较，结果表明本文提出的方法在大多数情况下表现更好。