Разработка алгоритма распознавания эмоций человека с использованием сверточной нейронной сети на основе аудиоданных

Objectives. This article provides a description and experience of creating the algorithm for recognizing the emotional state of the subject.Methods. Image processing methods are used.Results. The proposed algorithm makes it possible to recognize the emotional states of the subject on the basis of an audio data set. It was possible to improve the accuracy of the algorithm by changing the data set supplied to the input of the neural network.The stages of training convolutional neural network on a pre-prepared set of audio data are described, and the structure of the algorithm is described. To validate the neural network different set of audio data, not participating in the training, was selected. As a result of the study, graphs were constructed demonstrating the accuracy of the proposed method.After receiving the initial data of the study, the analysis of the possibilities for improving the algorithm in terms of ergonomics and accuracy of operation was also carried out. The strategy was developed to achieve a better result and obtain a more accurate algorithm. Based on the conclusions presented in the article, the rationale for choosing the representation of the data set and the software package necessary for the implementation of the software part of the algorithm is given.Conclusion. The proposed algorithm has a high accuracy of operation and does not require large computational costs.Цели. Приведено описание и рассмотрен опыт создания алгоритма распознавания эмоционального состояния субъекта.Методы. Использованы методы обработки изображений.Результаты. Предложенный алгоритм позволяет распознавать эмоциональные состояния субъекта на основании звукового набора данных. Благодаря проведенному исследованию удалось улучшить точность работы алгоритма путем изменения подаваемого на вход нейронной сети набора данных.Описаны этапы обучения сверточной нейронной сети на заранее заготовленном наборе звуковых данных, а также структура алгоритма. Для валидации нейронной сети был отобран иной, не участвующийв тренировке, набор аудиоданных. В результате проведения исследования построены графики, демонстрирующие точность работы предлагаемого метода.После получения первоначальных данных сделан анализ возможностей улучшения алгоритма с точки зрения эргономики и точности его работы. Разработана стратегия, позволяющая добиться лучшего результата и получить более точный алгоритм. На основании заключений, изложенных в статье, приводится обоснование выбора представления набора данных и программного комплекса, необходимого для реализации программной части алгоритма.Заключение. Предложенный алгоритм обладает высокой точностью и не требует больших вычислительных затрат

Attention-Inspired Artificial Neural Networks for Speech Processing: A Systematic Review

Artificial Neural Networks (ANNs) were created inspired by the neural networks in the human brain and have been widely applied in speech processing. The application areas of ANN include: Speech recognition, speech emotion recognition, language identification, speech enhancement, and speech separation, amongst others. Likewise, given that speech processing performed by humans involves complex cognitive processes known as auditory attention, there has been a growing amount of papers proposing ANNs supported by deep learning algorithms in conjunction with some mechanism to achieve symmetry with the human attention process. However, while these ANN approaches include attention, there is no categorization of attention integrated into the deep learning algorithms and their relation with human auditory attention. Therefore, we consider it necessary to have a review of the different ANN approaches inspired in attention to show both academic and industry experts the available models for a wide variety of applications. Based on the PRISMA methodology, we present a systematic review of the literature published since 2000, in which deep learning algorithms are applied to diverse problems related to speech processing. In this paper 133 research works are selected and the following aspects are described: (i) Most relevant features, (ii) ways in which attention has been implemented, (iii) their hypothetical relationship with human attention, and (iv) the evaluation metrics used. Additionally, the four publications most related with human attention were analyzed and their strengths and weaknesses were determined

An Experimental Analysis of Deep Learning Architectures for Supervised Speech Enhancement

Recent speech enhancement research has shown that deep learning techniques are very effective in removing background noise. Many deep neural networks are being proposed, showing promising results for improving overall speech perception. The Deep Multilayer Perceptron, Convolutional Neural Networks, and the Denoising Autoencoder are well-established architectures for speech enhancement; however, choosing between different deep learning models has been mainly empirical. Consequently, a comparative analysis is needed between these three architecture types in order to show the factors affecting their performance. In this paper, this analysis is presented by comparing seven deep learning models that belong to these three categories. The comparison includes evaluating the performance in terms of the overall quality of the output speech using five objective evaluation metrics and a subjective evaluation with 23 listeners; the ability to deal with challenging noise conditions; generalization ability; complexity; and, processing time. Further analysis is then provided while using two different approaches. The first approach investigates how the performance is affected by changing network hyperparameters and the structure of the data, including the Lombard effect. While the second approach interprets the results by visualizing the spectrogram of the output layer of all the investigated models, and the spectrograms of the hidden layers of the convolutional neural network architecture. Finally, a general evaluation is performed for supervised deep learning-based speech enhancement while using SWOC analysis, to discuss the technique’s Strengths, Weaknesses, Opportunities, and Challenges. The results of this paper contribute to the understanding of how different deep neural networks perform the speech enhancement task, highlight the strengths and weaknesses of each architecture, and provide recommendations for achieving better performance. This work facilitates the development of better deep neural networks for speech enhancement in the future

Automatic Identification of Emotional Information in Spanish TV Debates and Human-Machine Interactions

Automatic emotion detection is a very attractive field of research that can help build more natural human–machine interaction systems. However, several issues arise when real scenarios are considered, such as the tendency toward neutrality, which makes it difficult to obtain balanced datasets, or the lack of standards for the annotation of emotional categories. Moreover, the intrinsic subjectivity of emotional information increases the difficulty of obtaining valuable data to train machine learning-based algorithms. In this work, two different real scenarios were tackled: human–human interactions in TV debates and human–machine interactions with a virtual agent. For comparison purposes, an analysis of the emotional information was conducted in both. Thus, a profiling of the speakers associated with each task was carried out. Furthermore, different classification experiments show that deep learning approaches can be useful for detecting speakers’ emotional information, mainly for arousal, valence, and dominance levels, reaching a 0.7F1-score.The research presented in this paper was conducted as part of the AMIC and EMPATHIC projects, which received funding from the Spanish Minister of Science under grants TIN2017-85854-C4-3-R and PDC2021-120846-C43 and from the European Union’s Horizon 2020 research and innovation program under grant agreement No. 769872. The first author also received a PhD scholarship from the University of the Basque Country UPV/EHU, PIF17/310

Spatial Acoustic Vector Based Sound Field Reproduction

Spatial sound field reproduction aims to recreate an immersive sound field over a spatial region. The existing sound pressure based approaches to spatial sound field reproduction focus on the accurate approximation of original sound pressure over space, which ignores the perceptual accuracy of the reproduced sound field. The acoustic vectors of particle velocity and sound intensity appear to be closely linked with human perception of sound localization in literature. Therefore, in this thesis, we explore the spatial distributions of the acoustic vectors, and seek to develop algorithms to perceptually reproduce the original sound field over a continuous spatial region based on the vectors. A theory of spatial acoustic vectors is first developed, where the spatial distributions of particle velocity and sound intensity are derived from sound pressure. To extract the desired sound pressure from a mixed sound field environment, a 3D sound field separation technique is also formulated. Based on this theory, a series of reproduction techniques are proposed to improve the perceptual performance. The outcomes resulting from this theory are: (i) derivation of a particle velocity assisted 3D sound field reproduction technique which allows for non-uniform loudspeaker geometry with a limited number of loudspeakers, (ii) design of particle velocity based mixed-source sound field translation technique for binaural reproduction that can provide sound field translation with good perceptual experience over a large space, (iii) derivation of an intensity matching technique that can reproduce the desired sound field in a spherical region by controlling the sound intensity on the surface of the region, and (iv) two intensity based multizone sound field reproduction algorithms that can reproduce the desired sound field over multiple spatial zones. Finally, these techniques are evaluated by comparing to the conventional approaches through numerical simulations and real-world experiments

Classification of Sound Scenes and Events in Real-World Scenarios with Deep Learning Techniques

La clasificación de los eventos sonoros es un campo de la audición por computador que se está volviendo cada vez más interesante debido al gran número de aplicaciones que podrían beneficiarse de esta tecnología. A diferencia de otros campos de la audición por computador relacionados con la recuperación de información musical o el reconocimiento del habla, la clasificación de eventos sonoros tiene una serie de problemas intrínsecos. Estos problemas son la naturaleza polifónica de la mayoría de las grabaciones de sonido ambiental, la diferencia en la naturaleza de cada sonido, la falta de estructura temporal y la adición de ruido de fondo y reverberación en el proceso de grabación. Estos problemas son campos de estudio para la comunidad científica a día de hoy. Sin embargo, cabe señalar que cuando se despliega una solución de audición por computador en entornos reales, pueden surgir una serie de problemas adicionales. Estos problemas son el Reconocimiento de Conjunto Abierto (OSR), el Aprendizaje de Pocos Disparos (FSL) y la consideración del tiempo de ejecución del sistema (baja complejidad). El OSR se define como el problema que aparece cuando un sistema de inteligencia artificial tiene que enfrentarse a una situación desconocida en la que clases no vistas durante la etapa de entrenamiento están presentes en una etapa de inferencia. El FSL corresponde al problema que se produce cuando hay muy pocas muestras disponibles para cada clase considerada. Por último, dado que estos sistemas se despliegan normalmente en dispositivos de borde, hay que tener en cuenta el tiempo de ejecución, ya que cuanto menos tiempo tarde el sistema en dar una respuesta, mejor será la experiencia percibida por los usuarios. Las soluciones basadas en las técnicas de aprendizaje en profundidad para problemas similares en el dominio de la imagen han mostrado resultados prometedores. Las soluciones más difundidas son las que implementan Redes Neuronales Convolucionales (CNN). Por lo tanto, muchos sistemas de audio de última generación proponen convertir las señales de audio en una representación bidimensional que puede ser tratada como una imagen. La generación de mapas internos se realiza a menudo por las capas convolucionales de las CNN. Sin embargo, estas capas tienen una serie de limitaciones que deben ser estudiadas para poder proponer técnicas para mejorar los mapas de características resultantes. Con este fin, se han propuesto novedosas redes que fusionan dos métodos diferentes, como el aprendizaje residual y las técnicas de excitación y compresión. Los resultados muestran una mejora de la precisión del sistema con la adición de un número reducido de parámetros adicionales. Por otra parte, estas soluciones basadas en entradas bidimensionales pueden mostrar un cierto sesgo, ya que la elección de la representación de audio puede ser específica para una tarea concreta. Por lo tanto, se ha realizado un estudio comparativo de diferentes redes residuales alimentadas directamente por la señal de audio en bruto. Estas soluciones se conocen como de extremo a extremo. Si bien se han realizado estudios similares en la literatura en el dominio de la imagen, los resultados sugieren que los bloques residuales de mejor rendimiento para las tareas de visión artificial pueden no ser los mismos que los de la clasificación de audio. En cuanto a los problemas de FSL y OSR, se propone un marco basado en un autoencoder capaz de mitigar ambos problemas juntos. Esta solución es capaz de crear representaciones robustas de estos patrones de audio a partir de sólo unas pocas muestras, al tiempo que es capaz de rechazar las clases de audio no deseadas.The classification of sound events is a field of machine listening that is becoming increasingly interesting due to the large number of applications that could benefit from this technology. Unlike other fields of machine listening related to music information retrieval or speech recognition, sound event classification has a number of intrinsic problems. These problems are the polyphonic nature of most environmental sound recordings, the difference in the nature of each sound, the lack of temporal structure and the addition of background noise and reverberation in the recording process. These problems are fields of study for the scientific community today. However, it should be noted that when a machine listening solution is deployed in real environments, a number of extra problems may arise. These problems are Open-Set Recognition (OSR), Few-Shot Learning (FSL) and consideration of system runtime (low-complexity). OSR is defined as the problem that appears when an artificial intelligence system has to face an unknown situation where classes unseen during the training stage are present at a usage stage. FSL corresponds to the problem that occurs when there are very few samples available for each considered class. Finally, since these systems are normally deployed in edge devices, the consideration of the execution time must be taken into account, as the less time the system takes to give a response, the better the experience perceived by the users. Solutions based on Deep Learning techniques for similar problems in the image domain have shown promising results. The most widespread solutions are those that implement Convolutional Neural Networks (CNNs). Therefore, many state-of-the-art audio systems propose to convert audio signals into a two-dimensional representation that can be treated as an image. The generation of internal maps is often done by the convolutional layers of the CNNs. However, these layers have a series of limitations that must be studied in order to be able to propose techniques for improving the resulting feature maps. To this end, novel networks have been proposed that merge two different methods such as residual learning and squeeze-excitation techniques. The results show an improvement in the accuracy of the system with the addition of few number of extra parameters. On the other hand, these solutions based on two-dimensional inputs can show a certain bias since the choice of audio representation can be specific to a particular task. Therefore, a comparative study of different residual networks directly fed by the raw audio signal has been carried out. These solutions are known as end-to-end. While similar studies have been carried out in the literature in the image domain, the results suggest that the best performing residual blocks for computer vision tasks may not be the same as those for audio classification. Regarding the FSL and OSR problems, an autoencoder-based framework capable of mitigating both problems together is proposed. This solution is capable of creating robust representations of these audio patterns from just a few samples while being able to reject unwanted audio classes

Quality of experience in telemeetings and videoconferencing: a comprehensive survey

Telemeetings such as audiovisual conferences or virtual meetings play an increasingly important role in our professional and private lives. For that reason, system developers and service providers will strive for an optimal experience for the user, while at the same time optimizing technical and financial resources. This leads to the discipline of Quality of Experience (QoE), an active field originating from the telecommunication and multimedia engineering domains, that strives for understanding, measuring, and designing the quality experience with multimedia technology. This paper provides the reader with an entry point to the large and still growing field of QoE of telemeetings, by taking a holistic perspective, considering both technical and non-technical aspects, and by focusing on current and near-future services. Addressing both researchers and practitioners, the paper first provides a comprehensive survey of factors and processes that contribute to the QoE of telemeetings, followed by an overview of relevant state-of-the-art methods for QoE assessment. To embed this knowledge into recent technology developments, the paper continues with an overview of current trends, focusing on the field of eXtended Reality (XR) applications for communication purposes. Given the complexity of telemeeting QoE and the current trends, new challenges for a QoE assessment of telemeetings are identified. To overcome these challenges, the paper presents a novel Profile Template for characterizing telemeetings from the holistic perspective endorsed in this paper

Analysis and automatic identification of spontaneous emotions in speech from human-human and human-machine communication

383 p.This research mainly focuses on improving our understanding of human-human and human-machineinteractions by analysing paricipants¿ emotional status. For this purpose, we have developed andenhanced Speech Emotion Recognition (SER) systems for both interactions in real-life scenarios,explicitly emphasising the Spanish language. In this framework, we have conducted an in-depth analysisof how humans express emotions using speech when communicating with other persons or machines inactual situations. Thus, we have analysed and studied the way in which emotional information isexpressed in a variety of true-to-life environments, which is a crucial aspect for the development of SERsystems. This study aimed to comprehensively understand the challenge we wanted to address:identifying emotional information on speech using machine learning technologies. Neural networks havebeen demonstrated to be adequate tools for identifying events in speech and language. Most of themaimed to make local comparisons between some specific aspects; thus, the experimental conditions weretailored to each particular analysis. The experiments across different articles (from P1 to P19) are hardlycomparable due to our continuous learning of dealing with the difficult task of identifying emotions inspeech. In order to make a fair comparison, additional unpublished results are presented in the Appendix.These experiments were carried out under identical and rigorous conditions. This general comparisonoffers an overview of the advantages and disadvantages of the different methodologies for the automaticrecognition of emotions in speech