鉴于音乐源分离和自动混合的最新进展，在音乐曲目中删除音频效果是开发自动混合系统的有意义的一步。本文着重于消除对音乐制作中吉他曲目应用的失真音频效果。我们探索是否可以通过设计用于源分离和音频效应建模的神经网络来解决效果的去除。我们的方法证明对混合处理和清洁信号的效果特别有效。与基于稀疏优化的最新解决方案相比，这些模型获得了更好的质量和更快的推断。我们证明这些模型不仅适合倾斜，而且适用于其他类型的失真效应。通过讨论结果，我们强调了多个评估指标的有用性，以评估重建的不同方面的变形效果去除。

传统上，音乐混合涉及以干净，单个曲目的形式录制乐器，并使用音频效果和专家知识（例如，混合工程师）将它们融合到最终混合物中。近年来，音乐制作任务的自动化已成为一个新兴领域，基于规则的方法和机器学习方法已被探索。然而，缺乏干燥或干净的仪器记录限制了这种模型的性能，这与专业的人造混合物相去甚远。我们探索是否可以使用室外数据，例如潮湿或加工的多轨音乐录音，并将其重新利用以训练有监督的深度学习模型，以弥合自动混合质量的当前差距。为了实现这一目标，我们提出了一种新型的数据预处理方法，该方法允许模型执行自动音乐混合。我们还重新设计了一种用于评估音乐混合系统的听力测试方法。我们使用经验丰富的混合工程师作为参与者来验证结果。

Deep neural networks (DNN) techniques have become pervasive in domains such as natural language processing and computer vision. They have achieved great success in these domains in task such as machine translation and image generation. Due to their success, these data driven techniques have been applied in audio domain. More specifically, DNN models have been applied in speech enhancement domain to achieve denosing, dereverberation and multi-speaker separation in monaural speech enhancement. In this paper, we review some dominant DNN techniques being employed to achieve speech separation. The review looks at the whole pipeline of speech enhancement from feature extraction, how DNN based tools are modelling both global and local features of speech and model training (supervised and unsupervised). We also review the use of speech-enhancement pre-trained models to boost speech enhancement process. The review is geared towards covering the dominant trends with regards to DNN application in speech enhancement in speech obtained via a single speaker.

音乐源分离表示从给定歌曲中提取所有乐器的任务。近期对这一挑战的突破已经陷入了单一数据集，MusdB，仅限于四个仪器类。更大的数据集和更多乐器在收集数据和培训深度神经网络（DNN）时是昂贵和耗时的。在这项工作中，我们提出了一种快速的方法来评估任何数据集中的仪器在任何数据集中的可分离性，而不会训练和调整DNN。这种可分离性测量有助于选择适当的样本以获得神经网络的有效培训。基于Oracle原理与理想的比率面具，我们的方法是估计最先进的深度学习方法（如TASNet或Open-Unmix）的分离性能的优异代理。我们的结果有助于揭示音频源分离的两个基本要点：1）理想的比率掩模，虽然光线和简单，提供了最近神经网络的音频可分子性能的准确度量，以及2）新的端到端学习方法如TASNet，它直接在波形上运行，实际上是在内部构建时频（TF）表示，使得它们在分离在TF平面中重叠的音频模式时，它们遇到与基于TF的方法相同的限制。

最近在各种语音域应用中提出了卷积增强的变压器（构象异构体），例如自动语音识别（ASR）和语音分离，因为它们可以捕获本地和全球依赖性。在本文中，我们提出了一个基于构型的度量生成对抗网络（CMGAN），以在时间频率（TF）域中进行语音增强（SE）。发电机使用两阶段构象体块编码大小和复杂的频谱图信息，以模拟时间和频率依赖性。然后，解码器将估计分解为尺寸掩模的解码器分支，以滤除不需要的扭曲和复杂的细化分支，以进一步改善幅度估计并隐式增强相信息。此外，我们还包括一个度量歧视器来通过优化相应的评估评分来减轻度量不匹配。客观和主观评估表明，与三个语音增强任务（DeNoising，dereverberation和Super-Losity）中的最新方法相比，CMGAN能够表现出卓越的性能。例如，对语音库+需求数据集的定量降解分析表明，CMGAN的表现优于以前的差距，即PESQ为3.41，SSNR为11.10 dB。

我们提出了一个录音录音录音的录音录音。我们的模型通过短时傅立叶变换（STFT）将其输入转换为时频表示，并使用卷积神经网络处理所得的复杂频谱图。该网络在合成音乐数据集上培训了重建和对抗性目标，该数据集是通过将干净的音乐与从旧唱片的安静片段中提取的真实噪声样本混合而创建的。我们在合成数据集的持有测试示例中定量评估我们的方法，并通过人类对实际历史记录样本的评级进行定性评估。我们的结果表明，所提出的方法可有效消除噪音，同时保留原始音乐的质量和细节。

Objective: Despite numerous studies proposed for audio restoration in the literature, most of them focus on an isolated restoration problem such as denoising or dereverberation, ignoring other artifacts. Moreover, assuming a noisy or reverberant environment with limited number of fixed signal-to-distortion ratio (SDR) levels is a common practice. However, real-world audio is often corrupted by a blend of artifacts such as reverberation, sensor noise, and background audio mixture with varying types, severities, and duration. In this study, we propose a novel approach for blind restoration of real-world audio signals by Operational Generative Adversarial Networks (Op-GANs) with temporal and spectral objective metrics to enhance the quality of restored audio signal regardless of the type and severity of each artifact corrupting it. Methods: 1D Operational-GANs are used with generative neuron model optimized for blind restoration of any corrupted audio signal. Results: The proposed approach has been evaluated extensively over the benchmark TIMIT-RAR (speech) and GTZAN-RAR (non-speech) datasets corrupted with a random blend of artifacts each with a random severity to mimic real-world audio signals. Average SDR improvements of over 7.2 dB and 4.9 dB are achieved, respectively, which are substantial when compared with the baseline methods. Significance: This is a pioneer study in blind audio restoration with the unique capability of direct (time-domain) restoration of real-world audio whilst achieving an unprecedented level of performance for a wide SDR range and artifact types. Conclusion: 1D Op-GANs can achieve robust and computationally effective real-world audio restoration with significantly improved performance. The source codes and the generated real-world audio datasets are shared publicly with the research community in a dedicated GitHub repository1.

Single-channel, speaker-independent speech separation methods have recently seen great progress. However, the accuracy, latency, and computational cost of such methods remain insufficient. The majority of the previous methods have formulated the separation problem through the time-frequency representation of the mixed signal, which has several drawbacks, including the decoupling of the phase and magnitude of the signal, the suboptimality of time-frequency representation for speech separation, and the long latency in calculating the spectrograms. To address these shortcomings, we propose a fully-convolutional time-domain audio separation network (Conv-TasNet), a deep learning framework for end-to-end time-domain speech separation. Conv-TasNet uses a linear encoder to generate a representation of the speech waveform optimized for separating individual speakers. Speaker separation is achieved by applying a set of weighting functions (masks) to the encoder output. The modified encoder representations are then inverted back to the waveforms using a linear decoder. The masks are found using a temporal convolutional network (TCN) consisting of stacked 1-D dilated convolutional blocks, which allows the network to model the long-term dependencies of the speech signal while maintaining a small model size. The proposed Conv-TasNet system significantly outperforms previous time-frequency masking methods in separating two-and three-speaker mixtures. Additionally, Conv-TasNet surpasses several ideal time-frequency magnitude masks in two-speaker speech separation as evaluated by both objective distortion measures and subjective quality assessment by human listeners. Finally, Conv-TasNet has a significantly smaller model size and a shorter minimum latency, making it a suitable solution for both offline and real-time speech separation applications. This study therefore represents a major step toward the realization of speech separation systems for real-world speech processing technologies.

The marine ecosystem is changing at an alarming rate, exhibiting biodiversity loss and the migration of tropical species to temperate basins. Monitoring the underwater environments and their inhabitants is of fundamental importance to understand the evolution of these systems and implement safeguard policies. However, assessing and tracking biodiversity is often a complex task, especially in large and uncontrolled environments, such as the oceans. One of the most popular and effective methods for monitoring marine biodiversity is passive acoustics monitoring (PAM), which employs hydrophones to capture underwater sound. Many aquatic animals produce sounds characteristic of their own species; these signals travel efficiently underwater and can be detected even at great distances. Furthermore, modern technologies are becoming more and more convenient and precise, allowing for very accurate and careful data acquisition. To date, audio captured with PAM devices is frequently manually processed by marine biologists and interpreted with traditional signal processing techniques for the detection of animal vocalizations. This is a challenging task, as PAM recordings are often over long periods of time. Moreover, one of the causes of biodiversity loss is sound pollution; in data obtained from regions with loud anthropic noise, it is hard to separate the artificial from the fish sound manually. Nowadays, machine learning and, in particular, deep learning represents the state of the art for processing audio signals. Specifically, sound separation networks are able to identify and separate human voices and musical instruments. In this work, we show that the same techniques can be successfully used to automatically extract fish vocalizations in PAM recordings, opening up the possibility for biodiversity monitoring at a large scale.

从语音音频中删除背景噪音一直是大量研究和努力的主题，尤其是由于虚拟沟通和业余声音录制的兴起，近年来。然而，背景噪声并不是唯一可以防止可理解性的不愉快干扰：混响，剪裁，编解码器工件，有问题的均衡，有限的带宽或不一致的响度同样令人不安且无处不在。在这项工作中，我们建议将言语增强的任务视为一项整体努力，并提出了一种普遍的语音增强系统，同时解决了55种不同的扭曲。我们的方法由一种使用基于得分的扩散的生成模型以及一个多分辨率调节网络，该网络通过混合密度网络进行增强。我们表明，这种方法在专家听众执行的主观测试中大大优于艺术状态。我们还表明，尽管没有考虑任何特定的快速采样策略，但它仅通过4-8个扩散步骤就可以实现竞争性的目标得分。我们希望我们的方法论和技术贡献都鼓励研究人员和实践者采用普遍的语音增强方法，可能将其作为一项生成任务。

最近，基于扩散的生成模型已引入语音增强的任务。干净的语音损坏被建模为固定的远期过程，其中逐渐添加了越来越多的噪声。通过学习以嘈杂的输入为条件的迭代方式扭转这一过程，可以产生干净的语音。我们以先前的工作为基础，并在随机微分方程的形式主义中得出训练任务。我们对基础分数匹配目标进行了详细的理论综述，并探索了不同的采样器配置，以解决测试时的反向过程。通过使用自然图像生成文献的复杂网络体系结构，与以前的出版物相比，我们可以显着提高性能。我们还表明，我们可以与最近的判别模型竞争，并在评估与培训不同的语料库时获得更好的概括。我们通过主观的听力测试对评估结果进行补充，其中我们提出的方法是最好的。此外，我们表明所提出的方法在单渠道语音覆盖中实现了出色的最新性能。我们的代码和音频示例可在线获得，请参见https://uhh.de/inf-sp-sgmse

Music discovery services let users identify songs from short mobile recordings. These solutions are often based on Audio Fingerprinting, and rely more specifically on the extraction of spectral peaks in order to be robust to a number of distortions. Few works have been done to study the robustness of these algorithms to background noise captured in real environments. In particular, AFP systems still struggle when the signal to noise ratio is low, i.e when the background noise is strong. In this project, we tackle this problematic with Deep Learning. We test a new hybrid strategy which consists of inserting a denoising DL model in front of a peak-based AFP algorithm. We simulate noisy music recordings using a realistic data augmentation pipeline, and train a DL model to denoise them. The denoising model limits the impact of background noise on the AFP system's extracted peaks, improving its robustness to noise. We further propose a novel loss function to adapt the DL model to the considered AFP system, increasing its precision in terms of retrieved spectral peaks. To the best of our knowledge, this hybrid strategy has not been tested before.

使用多个麦克风进行语音增强的主要优点是，可以使用空间滤波来补充节奏光谱处理。在传统的环境中，通常单独执行线性空间滤波（波束形成）和单通道后过滤。相比之下，采用深层神经网络（DNN）有一种趋势来学习联合空间和速度 - 光谱非线性滤波器，这意味着对线性处理模型的限制以及空间和节奏单独处理的限制光谱信息可能可以克服。但是，尚不清楚导致此类数据驱动的过滤器以良好性能进行多通道语音增强的内部机制。因此，在这项工作中，我们通过仔细控制网络可用的信息源（空间，光谱和时间）来分析由DNN实现的非线性空间滤波器的性质及其与时间和光谱处理的相互依赖性。我们确认了非线性空间处理模型的优越性，该模型在挑战性的扬声器提取方案中优于Oracle线性空间滤波器，以低于0.24的POLQA得分，较少数量的麦克风。我们的分析表明，在特定的光谱信息中应与空间信息共同处理，因为这会提高过滤器的空间选择性。然后，我们的系统评估会导致一个简单的网络体系结构，该网络体系结构在扬声器提取任务上的最先进的网络体系结构优于0.22 POLQA得分，而CHIME3数据上的POLQA得分为0.32。

尽管近年来取得了惊人的进步，但最先进的音乐分离系统会产生具有显着感知缺陷的源估计，例如增加无关噪声或消除谐波。我们提出了一个后处理模型（MAKE听起来不错（MSG）后处理器），以增强音乐源分离系统的输出。我们将我们的后处理模型应用于最新的基于波形和基于频谱图的音乐源分离器，包括在训练过程中未见的分离器。我们对源分离器产生的误差的分析表明，波形模型倾向于引入更多高频噪声，而频谱图模型倾向于丢失瞬变和高频含量。我们引入了客观措施来量化这两种错误并显示味精改善了两种错误的源重建。众包主观评估表明，人类的听众更喜欢由MSG进行后处理的低音和鼓的来源估计。

生成的对抗网络最近在神经声音中表现出了出色的表现，表现优于最佳自动回归和基于流动的模型。在本文中，我们表明这种成功可以扩展到有条件音频的其他任务。特别是，在HIFI Vocoders的基础上，我们为带宽扩展和语音增强的新型HIFI ++一般框架提出了新颖的一般框架。我们表明，通过改进的生成器体系结构和简化的多歧视培训，HIFI ++在这些任务中的最先进的情况下表现更好或与之相提并论，同时花费大量的计算资源。通过一系列广泛的实验，我们的方法的有效性得到了验证。

隔离架构在语音分离中显示出非常好的结果。像其他学习的编码器模型一样，它使用了短帧，因为它们已被证明在这些情况下可以获得更好的性能。这导致输入处有大量帧，这是有问题的。由于隔离器是基于变压器的，因此其计算复杂性随着较长的序列而大大增加。在本文中，我们在语音增强任务中采用了隔离器，并表明，通过以短期傅立叶变换（STFT）表示替换学习式编码器的功能，我们可以使用长帧而不会损害感知增强性能。我们获得了同等的质量和清晰度评估得分，同时将10秒的话语减少了大约8倍。

设备方向听到需要从给定方向的音频源分离，同时实现严格的人类难以察觉的延迟要求。虽然神经网络可以实现比传统的波束形成器的性能明显更好，但所有现有型号都缺乏对计算受限的可穿戴物的低延迟因果推断。我们展示了一个混合模型，将传统的波束形成器与定制轻质神经网络相结合。前者降低了后者的计算负担，并且还提高了其普遍性，而后者旨在进一步降低存储器和计算开销，以实现实时和低延迟操作。我们的评估显示了合成数据上最先进的因果推断模型的相当性能，同时实现了模型尺寸的5倍，每秒计算的4倍，处理时间减少5倍，更好地概括到真实的硬件数据。此外，我们的实时混合模型在为低功耗可穿戴设备设计的移动CPU上运行8毫秒，并实现17.5毫秒的端到端延迟。

我们提出了一个单阶段的休闲波形到波形多通道模型，该模型可以根据动态的声学场景中的广泛空间位置分离移动的声音源。我们将场景分为两个空间区域，分别包含目标和干扰声源。该模型经过训练有素的端到端，并隐含地进行空间处理，而没有基于传统处理或使用手工制作的空间特征的任何组件。我们在现实世界数据集上评估了所提出的模型，并表明该模型与Oracle Beamformer的性能匹配，然后是最先进的单渠道增强网络。

最近的单声道源分离的工作表明，通过使用短窗户的完全学习过滤器组可以提高性能。另一方面，广泛众所周知，对于传统的波束成形技术，性能随着长分析窗口而增加。这也适用于最依赖于深神经网络（DNN）来估计空间协方差矩阵的大多数混合神经波束形成方法。在这项工作中，我们尝试弥合这两个世界之间的差距，并探索完全端到端的混合神经波束形成，而不是使用短时傅里叶变换，而不是使用DNN共同学习分析和合成滤波器拦截器。详细说明，我们探索了两种不同类型的学习过滤博客：完全学习和分析。我们使用最近的清晰度挑战数据执行详细分析，并显示通过使用学习的默认覆盖机，可以超越基于Oracle掩码的短窗口的波束成形。

Current audio-visual separation methods share a standard architecture design where an audio encoder-decoder network is fused with visual encoding features at the encoder bottleneck. This design confounds the learning of multi-modal feature encoding with robust sound decoding for audio separation. To generalize to a new instrument: one must finetune the entire visual and audio network for all musical instruments. We re-formulate visual-sound separation task and propose Instrument as Query (iQuery) with a flexible query expansion mechanism. Our approach ensures cross-modal consistency and cross-instrument disentanglement. We utilize "visually named" queries to initiate the learning of audio queries and use cross-modal attention to remove potential sound source interference at the estimated waveforms. To generalize to a new instrument or event class, drawing inspiration from the text-prompt design, we insert an additional query as an audio prompt while freezing the attention mechanism. Experimental results on three benchmarks demonstrate that our iQuery improves audio-visual sound source separation performance.