Nonlinear factor analysis and its application to acoustical source separation and identification

Acoustical signals of mechanical systems can provide original information of operating conditions, and thus benefit for machinery condition monitoring and fault diagnosis. However, acoustical signals measured by sensors are mixed signals of all the sources, and normally it is impossible to be directly used for acoustical source identification or feature extraction. Therefore, this paper presents nonlinear factor analysis (NLFA) and applies it to acoustical source separation and identification of mechanical systems. The effects by numbers of hidden neurons and mixed signals on separation performances of NLFA are comparatively studied. Furthermore, acoustical signals from a test bed with shell structures are separated and identified by NLFA and correlation analysis, and the effectiveness of NLFA on acoustical signals is validated by both numerical case studies and an experimental case study. This work can benefit for machinery noise monitoring, reduction and control, and also provide pure source information for machinery condition monitoring or fault diagnosis

Iterative Separation of Note Events from Single-Channel Polyphonic Recordings

This thesis is concerned with the separation of audio sources from single-channel polyphonic musical recordings using the iterative estimation and separation of note events. Each event is defined as a section of audio containing largely harmonic energy identified as coming from a single sound source. Multiple events can be clustered to form separated sources. This solution is a model-based algorithm that can be applied to a large variety of audio recordings without requiring previous training stages. The proposed system embraces two principal stages. The first one considers the iterative detection and separation of note events from within the input mixture. In every iteration, the pitch trajectory of the predominant note event is automatically selected from an array of fundamental frequency estimates and used to guide the separation of the event's spectral content using two different methods: time-frequency masking and time-domain subtraction. A residual signal is then generated and used as the input mixture for the next iteration. After convergence, the second stage considers the clustering of all detected note events into individual audio sources. Performance evaluation is carried out at three different levels. Firstly, the accuracy of the note-event-based multipitch estimator is compared with that of the baseline algorithm used in every iteration to generate the initial set of pitch estimates. Secondly, the performance of the semi-supervised source separation process is compared with that of another semi-automatic algorithm. Finally, a listening test is conducted to assess the audio quality and naturalness of the separated sources when they are used to create stereo mixes from monaural recordings. Future directions for this research focus on the application of the proposed system to other music-related tasks. Also, a preliminary optimisation-based approach is presented as an alternative method for the separation of overlapping partials, and as a high resolution time-frequency representation for digital signals

Detecting multiple, simultaneous talkers through localising speech recorded by ad-hoc microphone arrays

This paper proposes a novel approach to detecting multiple, simultaneous talkers in multi-party meetings using localisation of active speech sources recorded with an ad-hoc microphone array. Cues indicating the relative distance between sources and microphones are derived from speech signals and room impulse responses recorded by each of the microphones distributed at unknown locations within a room. Multiple active sources are localised by analysing a surface formed from these cues and derived at different locations within the room. The number of localised active sources per each frame or utterance is then counted to estimate when multiple sources are active. The proposed approach does not require prior information about the number and locations of sources or microphones. Synchronisation between microphones is also not required. A meeting scenario with competing speakers is simulated and results show that simultaneously active sources can be detected with an average accuracy of 75% and the number of active sources counted accurately 65% of the time

Audio source separation for music in low-latency and high-latency scenarios

Aquesta tesi proposa mètodes per tractar les limitacions de les tècniques existents de separació de fonts musicals en condicions de baixa i alta latència. En primer lloc, ens centrem en els mètodes amb un baix cost computacional i baixa latència. Proposem l'ús de la regularització de Tikhonov com a mètode de descomposició de l'espectre en el context de baixa latència. El comparem amb les tècniques existents en tasques d'estimació i seguiment dels tons, que són passos crucials en molts mètodes de separació. A continuació utilitzem i avaluem el mètode de descomposició de l'espectre en tasques de separació de veu cantada, baix i percussió. En segon lloc, proposem diversos mètodes d'alta latència que milloren la separació de la veu cantada, gràcies al modelatge de components específics, com la respiració i les consonants. Finalment, explorem l'ús de correlacions temporals i anotacions manuals per millorar la separació dels instruments de percussió i dels senyals musicals polifònics complexes.Esta tesis propone métodos para tratar las limitaciones de las técnicas existentes de separación de fuentes musicales en condiciones de baja y alta latencia. En primer lugar, nos centramos en los métodos con un bajo coste computacional y baja latencia. Proponemos el uso de la regularización de Tikhonov como método de descomposición del espectro en el contexto de baja latencia. Lo comparamos con las técnicas existentes en tareas de estimación y seguimiento de los tonos, que son pasos cruciales en muchos métodos de separación. A continuación utilizamos y evaluamos el método de descomposición del espectro en tareas de separación de voz cantada, bajo y percusión. En segundo lugar, proponemos varios métodos de alta latencia que mejoran la separación de la voz cantada, gracias al modelado de componentes que a menudo no se toman en cuenta, como la respiración y las consonantes. Finalmente, exploramos el uso de correlaciones temporales y anotaciones manuales para mejorar la separación de los instrumentos de percusión y señales musicales polifónicas complejas.This thesis proposes specific methods to address the limitations of current music source separation methods in low-latency and high-latency scenarios. First, we focus on methods with low computational cost and low latency. We propose the use of Tikhonov regularization as a method for spectrum decomposition in the low-latency context. We compare it to existing techniques in pitch estimation and tracking tasks, crucial steps in many separation methods. We then use the proposed spectrum decomposition method in low-latency separation tasks targeting singing voice, bass and drums. Second, we propose several high-latency methods that improve the separation of singing voice by modeling components that are often not accounted for, such as breathiness and consonants. Finally, we explore using temporal correlations and human annotations to enhance the separation of drums and complex polyphonic music signals

Proceedings of the EAA Spatial Audio Signal Processing symposium: SASP 2019

International audienc

Rétroingénierie du son pour l écoute active et autres applications

Ce travail s intéresse au problème de la rétroingénierie du son pour l écoute active. Le format considéré correspond au CD audio. Le contenu musical est vu comme le résultat d un enchaînement de la composition, l enregistrement, le mixage et le mastering. L inversion des deux dernières étapes constitue le fond du problème présent. Le signal audio est traité comme un mélange post-non-linéaire. Ainsi, le mélange est décompressé avant d'être décomposé en pistes audio. Le problème est abordé dans un contexte informé : l inversion est accompagnée d'une information qui est spécifique à la production du contenu. De cette manière, la qualité de l inversion est significativement améliorée. L information est réduite de taille en se servant des méthodes de quantification, codage, et des faits sur la psychoacoustique. Les méthodes proposées s appliquent en temps réel et montrent une complexité basse. Les résultats obtenus améliorent l état de l art et contribuent aux nouvelles connaissances.This work deals with the problem of reverse audio engineering for active listening. The format under consideration corresponds to the audio CD. The musical content is viewed as the result of a concatenation of the composition, the recording, the mixing, and the mastering. The inversion of the two latter stages constitutes the core of the problem at hand. The audio signal is treated as a post-nonlinear mixture. Thus, the mixture is decompressed before being decomposed into audio tracks. The problem is tackled in an informed context: The inversion is accompanied by information which is specific to the content production. In this manner, the quality of the inversion is significantly improved. The information is reduced in size by the use of quantification and coding methods, and some facts on psychoacoustics. The proposed methods are applicable in real time and have a low complexity. The obtained results advance the state of the art and contribute new insights.BORDEAUX1-Bib.electronique (335229901) / SudocSudocFranceF

Principled methods for mixtures processing

This document is my thesis for getting the habilitation à diriger des recherches, which is the french diploma that is required to fully supervise Ph.D. students. It summarizes the research I did in the last 15 years and also provides the short­term research directions and applications I want to investigate. Regarding my past research, I first describe the work I did on probabilistic audio modeling, including the separation of Gaussian and α­stable stochastic processes. Then, I mention my work on deep learning applied to audio, which rapidly turned into a large effort for community service. Finally, I present my contributions in machine learning, with some works on hardware compressed sensing and probabilistic generative models.My research programme involves a theoretical part that revolves around probabilistic machine learning, and an applied part that concerns the processing of time series arising in both audio and life sciences

The creation of a binaural spatialization tool

The main focus of the research presented within this thesis is, as the title suggests, binaural spatialization. Binaural technology and, especially, the binaural recording technique are not particu-larly recent. Nevertheless, the interest in this technology has lately become substantial due to the increase in the calculation power of personal computers, which started to allow the complete and accurate real-time simulation of three-dimensional sound-fields over headphones. The goals of this body of research have been determined in order to provide elements of novelty and of contribution to the state of the art in the field of binaural spatialization. A brief summary of these is found in the following list: • The development and implementation of a binaural spatialization technique with Distance Simulation, based on the individual simulation of the distance cues and Binaural Reverb, in turn based on the weighted mix between the signals convolved with the different HRIR and BRIR sets; • The development and implementation of a characterization process for modifying a BRIR set in order to simulate different environments with different characteristics in terms of frequency response and reverb time; • The creation of a real-time and offline binaural spatialization application, imple-menting the techniques cited in the previous points, and including a set of multichannel(and Ambisonics)-to-binaural conversion tools. • The performance of a perceptual evaluation stage to verify the effectiveness, realism, and quality of the techniques developed, and • The application and use of the developed tools within both scientific and artistic “case studies”. In the following chapters, sections, and subsections, the research performed between January 2006 and March 2010 will be described, outlining the different stages before, during, and after the development of the software platform, analysing the results of the perceptual evaluations and drawing conclusions that could, in the future, be considered the starting point for new and innovative research projects

Movements in Binaural Space: Issues in HRTF Interpolation and Reverberation, with applications to Computer Music

This thesis deals broadly with the topic of Binaural Audio. After reviewing the literature, a reappraisal of the minimum-phase plus linear delay model for HRTF representation and interpolation is offered. A rigorous analysis of threshold based phase unwrapping is also performed. The results and conclusions drawn from these analyses motivate the development of two novel methods for HRTF representation and interpolation. Empirical data is used directly in a Phase Truncation method. A Functional Model for phase is used in the second method based on the psychoacoustical nature of Interaural Time Differences. Both methods are validated; most significantly, both perform better than a minimum-phase method in subjective testing. The accurate, artefact-free dynamic source processing afforded by the above methods is harnessed in a binaural reverberation model, based on an early reflection image model and Feedback Delay Network diffuse field, with accurate interaural coherence. In turn, these flexible environmental processing algorithms are used in the development of a multi-channel binaural application, which allows the audition of multi-channel setups in headphones. Both source and listener are dynamic in this paradigm. A GUI is offered for intuitive use of the application. HRTF processing is thus re-evaluated and updated after a review of accepted practice. Novel solutions are presented and validated. Binaural reverberation is recognised as a crucial tool for convincing artificial spatialisation, and is developed on similar principles. Emphasis is placed on transparency of development practices, with the aim of wider dissemination and uptake of binaural technology

Sonic interactions in virtual environments

This book tackles the design of 3D spatial interactions in an audio-centered and audio-first perspective, providing the fundamental notions related to the creation and evaluation of immersive sonic experiences. The key elements that enhance the sensation of place in a virtual environment (VE) are: Immersive audio: the computational aspects of the acoustical-space properties of Virutal Reality (VR) technologies Sonic interaction: the human-computer interplay through auditory feedback in VE VR systems: naturally support multimodal integration, impacting different application domains Sonic Interactions in Virtual Environments will feature state-of-the-art research on real-time auralization, sonic interaction design in VR, quality of the experience in multimodal scenarios, and applications. Contributors and editors include interdisciplinary experts from the fields of computer science, engineering, acoustics, psychology, design, humanities, and beyond. Their mission is to shape an emerging new field of study at the intersection of sonic interaction design and immersive media, embracing an archipelago of existing research spread in different audio communities and to increase among the VR communities, researchers, and practitioners, the awareness of the importance of sonic elements when designing immersive environments