Semi-supervised Multi-modal Emotion Recognition with Cross-Modal Distribution Matching

Automatic emotion recognition is an active research topic with wide range of applications. Due to the high manual annotation cost and inevitable label ambiguity, the development of emotion recognition dataset is limited in both scale and quality. Therefore, one of the key challenges is how to build effective models with limited data resource. Previous works have explored different approaches to tackle this challenge including data enhancement, transfer learning, and semi-supervised learning etc. However, the weakness of these existing approaches includes such as training instability, large performance loss during transfer, or marginal improvement. In this work, we propose a novel semi-supervised multi-modal emotion recognition model based on cross-modality distribution matching, which leverages abundant unlabeled data to enhance the model training under the assumption that the inner emotional status is consistent at the utterance level across modalities. We conduct extensive experiments to evaluate the proposed model on two benchmark datasets, IEMOCAP and MELD. The experiment results prove that the proposed semi-supervised learning model can effectively utilize unlabeled data and combine multi-modalities to boost the emotion recognition performance, which outperforms other state-of-the-art approaches under the same condition. The proposed model also achieves competitive capacity compared with existing approaches which take advantage of additional auxiliary information such as speaker and interaction context.Comment: 10 pages, 5 figures, to be published on ACM Multimedia 202

Speech emotion recognition based on bi-directional acoustic–articulatory conversion

Acoustic and articulatory signals are naturally coupled and complementary. The challenge of acquiring articulatory data and the nonlinear ill-posedness of acoustic–articulatory conversions have resulted in previous studies on speech emotion recognition (SER) primarily relying on unidirectional acoustic–articulatory conversions. However, these studies have ignored the potential benefits of bi-directional acoustic–articulatory conversion. Addressing the problem of nonlinear ill-posedness and effectively extracting and utilizing these two modal features in SER remain open research questions. To bridge this gap, this study proposes a Bi-A2CEmo framework that simultaneously addresses the bi-directional acoustic-articulatory conversion for SER. This framework comprises three components: a Bi-MGAN that addresses the nonlinear ill-posedness problem, KCLNet that enhances the emotional attributes of the mapped features, and ResTCN-FDA that fully exploits the emotional attributes of the features. Another challenge is the absence of a parallel acoustic-articulatory emotion database. To overcome this issue, this study utilizes electromagnetic articulography (EMA) to create a multi-modal acoustic-articulatory emotion database for Mandarin Chinese called STEM-E$^2$VA. A comparative analysis is then conducted between the proposed method and state-of-the-art models to evaluate the effectiveness of the framework. Bi-A2CEmo achieves an accuracy of 89.04\% in SER, which is an improvement of 5.27\% compared with the actual acoustic and articulatory features recorded by the EMA. The results for the STEM-E$^2$VA dataset show that Bi-MGAN achieves a higher accuracy in mapping and inversion than conventional conversion networks. Visualization of the mapped features before and after enhancement reveals that KCLNet reduces the intra-class spacing while increasing the inter-class spacing of the features. ResTCN-FDA demonstrates high recognition accuracy on three publicly available datasets. The experimental results show that the proposed bi-directional acoustic-articulatory conversion framework can significantly improve the SER performance

Audio Signal Enhancement with Learning from Positive and Unlabelled Data

Supervised learning is a mainstream approach to audio signal enhancement (SE) and requires parallel training data consisting of both noisy signals and the corresponding clean signals. Such data can only be synthesised and are mismatched with real data, which can result in poor performance on real data. Moreover, clean signals may be inaccessible in certain scenarios, which renders this conventional approach infeasible. Here we explore SE using non-parallel training data consisting of noisy signals and noise, which can be easily recorded. We define the positive (P) and the negative (N) classes as signal inactivity and activity, respectively. We observe that the spectrogram patches of noise clips can be used as P data and those of noisy signal clips as unlabelled data. Thus, learning from positive and unlabelled data enables a convolutional neural network to learn to classify each spectrogram patch as P or N to enable SE.Comment: Accepted to ICASSP202

A Review of Deep Learning Techniques for Speech Processing

The field of speech processing has undergone a transformative shift with the advent of deep learning. The use of multiple processing layers has enabled the creation of models capable of extracting intricate features from speech data. This development has paved the way for unparalleled advancements in speech recognition, text-to-speech synthesis, automatic speech recognition, and emotion recognition, propelling the performance of these tasks to unprecedented heights. The power of deep learning techniques has opened up new avenues for research and innovation in the field of speech processing, with far-reaching implications for a range of industries and applications. This review paper provides a comprehensive overview of the key deep learning models and their applications in speech-processing tasks. We begin by tracing the evolution of speech processing research, from early approaches, such as MFCC and HMM, to more recent advances in deep learning architectures, such as CNNs, RNNs, transformers, conformers, and diffusion models. We categorize the approaches and compare their strengths and weaknesses for solving speech-processing tasks. Furthermore, we extensively cover various speech-processing tasks, datasets, and benchmarks used in the literature and describe how different deep-learning networks have been utilized to tackle these tasks. Additionally, we discuss the challenges and future directions of deep learning in speech processing, including the need for more parameter-efficient, interpretable models and the potential of deep learning for multimodal speech processing. By examining the field's evolution, comparing and contrasting different approaches, and highlighting future directions and challenges, we hope to inspire further research in this exciting and rapidly advancing field

Sound-to-imagination: an exploratory study on cross-modal translation using diverse audiovisual data

The motivation of our research is to explore the possibilities of automatic sound-to-image (S2I) translation for enabling a human receiver to visually infer occurrences of sound-related events. We expect the computer to ‘imagine’ scenes from captured sounds, generating original images that depict the sound-emitting sources. Previous studies on similar topics opted for simplified approaches using data with low content diversity and/or supervision/self-supervision for training. In contrast, our approach involves performing S2I translation using thousands of distinct and unknown scenes, using sound class annotations solely for data preparation, just enough to ensure aural–visual semantic coherence. To model the translator, we employ an audio encoder and a conditional generative adversarial network (GAN) with a deep densely connected generator. Furthermore, we present a solution using informativity classifiers for quantitatively evaluating the generated images. This allows us to analyze the influence of network-bottleneck variation on the translation process, highlighting a potential trade-off between informativity and pixel space convergence. Despite the complexity of the specified S2I translation task, we were able to generalize the model enough to obtain more than 14%, on average, of interpretable and semantically coherent images translated from unknown sounds.The present work was supported in part by the Brazilian National Council for Scientific and Technological Development (CNPq) under PhD grant 200884/2015-8. Also, the work was partly supported by the Spanish State Research Agency (AEI), project PID2019-107579RBI00/AEI/10.13039/501100011033.Peer ReviewedPostprint (published version

Scaling Machine Learning Systems using Domain Adaptation

Machine-learned components, particularly those trained using deep learning methods, are becoming integral parts of modern intelligent systems, with applications including computer vision, speech processing, natural language processing and human activity recognition. As these machine learning (ML) systems scale to real-world settings, they will encounter scenarios where the distribution of the data in the real-world (i.e., the target domain) is different from the data on which they were trained (i.e., the source domain). This phenomenon, known as domain shift, can significantly degrade the performance of ML systems in new deployment scenarios. In this thesis, we study the impact of domain shift caused by variations in system hardware, software and user preferences on the performance of ML systems. After quantifying the performance degradation of ML models in target domains due to the various types of domain shift, we propose unsupervised domain adaptation (uDA) algorithms that leverage unlabeled data collected in the target domain to improve the performance of the ML model. At its core, this thesis argues for the need to develop uDA solutions while adhering to practical scenarios in which ML systems will scale. More specifically, we consider four scenarios: (i) opaque ML systems, wherein parameters of the source prediction model are not made accessible in the target domain, (ii) transparent ML systems, wherein source model parameters are accessible and can be modified in the target domain, (iii) ML systems where source and target domains do not have identical label spaces, and (iv) distributed ML systems, wherein the source and target domains are geographically distributed, their datasets are private and cannot be exchanged using adaptation. We study the unique challenges and constraints of each scenario and propose novel uDA algorithms that outperform state-of-the-art baselines