Video-aided model-based source separation in real reverberant rooms

Source separation algorithms that utilize only audio data can perform poorly if multiple sources or reverberation are present. In this paper we therefore propose a video-aided model-based source separation algorithm for a two-channel reverberant recording in which the sources are assumed static. By exploiting cues from video, we first localize individual speech sources in the enclosure and then estimate their directions. The interaural spatial cues, the interaural phase difference and the interaural level difference, as well as the mixing vectors are probabilistically modeled. The models make use of the source direction information and are evaluated at discrete timefrequency points. The model parameters are refined with the wellknown expectation-maximization (EM) algorithm. The algorithm outputs time-frequency masks that are used to reconstruct the individual sources. Simulation results show that by utilizing the visual modality the proposed algorithm can produce better timefrequency masks thereby giving improved source estimates. We provide experimental results to test the proposed algorithm in different scenarios and provide comparisons with both other audio-only and audio-visual algorithms and achieve improved performance both on synthetic and real data. We also include dereverberation based pre-processing in our algorithm in order to suppress the late reverberant components from the observed stereo mixture and further enhance the overall output of the algorithm. This advantage makes our algorithm a suitable candidate for use in under-determined highly reverberant settings where the performance of other audio-only and audio-visual methods is limited

Effect of Head Movement on Sound Localization in Real and Simulated Cochlear Implant Users

Cochlear implant (CI) users’ limited ability to use acoustical cues for sound localization causes left/right confusions and front/back reversals. Head movement is beneficial in reducing these errors in acoustically hearing listeners. This study investigated the effect of head movement on localization throughout 360o of azimuth for both real and simulated CI users. Listeners in a bilateral electro-acoustic (CI with ipsilateral hearing aid) simulation derived the greatest head movement benefit in reducing front/back reversals. Left/right confusions were reduced in simulations with matched bilateral stimulation. Sensitivity to both timing and level cues for sound localization was correlated with sound localization performance without head movement for simulated device users. Sensitivity to timing cues was correlated with sound localization performance with head movement cues for simulated device users. Simulations of bilateral CI and bimodal users (CI with contralateral hearing aid) listening predicted real users’ sound localization performance, binaural sensitivity and head movement patterns

Sound Event Localization, Detection, and Tracking by Deep Neural Networks

In this thesis, we present novel sound representations and classification methods for the task of sound event localization, detection, and tracking (SELDT). The human auditory system has evolved to localize multiple sound events, recognize and further track their motion individually in an acoustic environment. This ability of humans makes them context-aware and enables them to interact with their surroundings naturally. Developing similar methods for machines will provide an automatic description of social and human activities around them and enable machines to be context-aware similar to humans. Such methods can be employed to assist the hearing impaired to visualize sounds, for robot navigation, and to monitor biodiversity, the home, and cities. A real-life acoustic scene is complex in nature, with multiple sound events that are temporally and spatially overlapping, including stationary and moving events with varying angular velocities. Additionally, each individual sound event class, for example, a car horn can have a lot of variabilities, i.e., different cars have different horns, and within the same model of the car, the duration and the temporal structure of the horn sound is driver dependent. Performing SELDT in such overlapping and dynamic sound scenes while being robust is challenging for machines. Hence we propose to investigate the SELDT task in this thesis and use a data-driven approach using deep neural networks (DNNs). The sound event detection (SED) task requires the detection of onset and offset time for individual sound events and their corresponding labels. In this regard, we propose to use spatial and perceptual features extracted from multichannel audio for SED using two different DNNs, recurrent neural networks (RNNs) and convolutional recurrent neural networks (CRNNs). We show that using multichannel audio features improves the SED performance for overlapping sound events in comparison to traditional single-channel audio features. The proposed novel features and methods produced state-of-the-art performance for the real-life SED task and won the IEEE AASP DCASE challenge consecutively in 2016 and 2017. Sound event localization is the task of spatially locating the position of individual sound events. Traditionally, this has been approached using parametric methods. In this thesis, we propose a CRNN for detecting the azimuth and elevation angles of multiple temporally overlapping sound events. This is the first DNN-based method performing localization in complete azimuth and elevation space. In comparison to parametric methods which require the information of the number of active sources, the proposed method learns this information directly from the input data and estimates their respective spatial locations. Further, the proposed CRNN is shown to be more robust than parametric methods in reverberant scenarios. Finally, the detection and localization tasks are performed jointly using a CRNN. This method additionally tracks the spatial location with time, thus producing the SELDT results. This is the first DNN-based SELDT method and is shown to perform equally with stand-alone baselines for SED, localization, and tracking. The proposed SELDT method is evaluated on nine datasets that represent anechoic and reverberant sound scenes, stationary and moving sources with varying velocities, a different number of overlapping sound events and different microphone array formats. The results show that the SELDT method can track multiple overlapping sound events that are both spatially stationary and moving

Binaural scene analysis : localization, detection and recognition of speakers in complex acoustic scenes

The human auditory system has the striking ability to robustly localize and recognize a specific target source in complex acoustic environments while ignoring interfering sources. Surprisingly, this remarkable capability, which is referred to as auditory scene analysis, is achieved by only analyzing the waveforms reaching the two ears. Computers, however, are presently not able to compete with the performance achieved by the human auditory system, even in the restricted paradigm of confronting a computer algorithm based on binaural signals with a highly constrained version of auditory scene analysis, such as localizing a sound source in a reverberant environment or recognizing a speaker in the presence of interfering noise. In particular, the problem of focusing on an individual speech source in the presence of competing speakers, termed the cocktail party problem, has been proven to be extremely challenging for computer algorithms. The primary objective of this thesis is the development of a binaural scene analyzer that is able to jointly localize, detect and recognize multiple speech sources in the presence of reverberation and interfering noise. The processing of the proposed system is divided into three main stages: localization stage, detection of speech sources, and recognition of speaker identities. The only information that is assumed to be known a priori is the number of target speech sources that are present in the acoustic mixture. Furthermore, the aim of this work is to reduce the performance gap between humans and machines by improving the performance of the individual building blocks of the binaural scene analyzer. First, a binaural front-end inspired by auditory processing is designed to robustly determine the azimuth of multiple, simultaneously active sound sources in the presence of reverberation. The localization model builds on the supervised learning of azimuthdependent binaural cues, namely interaural time and level differences. Multi-conditional training is performed to incorporate the uncertainty of these binaural cues resulting from reverberation and the presence of competing sound sources. Second, a speech detection module that exploits the distinct spectral characteristics of speech and noise signals is developed to automatically select azimuthal positions that are likely to correspond to speech sources. Due to the established link between the localization stage and the recognition stage, which is realized by the speech detection module, the proposed binaural scene analyzer is able to selectively focus on a predefined number of speech sources that are positioned at unknown spatial locations, while ignoring interfering noise sources emerging from other spatial directions. Third, the speaker identities of all detected speech sources are recognized in the final stage of the model. To reduce the impact of environmental noise on the speaker recognition performance, a missing data classifier is combined with the adaptation of speaker models using a universal background model. This combination is particularly beneficial in nonstationary background noise

Online Localization and Tracking of Multiple Moving Speakers in Reverberant Environments

We address the problem of online localization and tracking of multiple moving speakers in reverberant environments. The paper has the following contributions. We use the direct-path relative transfer function (DP-RTF), an inter-channel feature that encodes acoustic information robust against reverberation, and we propose an online algorithm well suited for estimating DP-RTFs associated with moving audio sources. Another crucial ingredient of the proposed method is its ability to properly assign DP-RTFs to audio-source directions. Towards this goal, we adopt a maximum-likelihood formulation and we propose to use an exponentiated gradient (EG) to efficiently update source-direction estimates starting from their currently available values. The problem of multiple speaker tracking is computationally intractable because the number of possible associations between observed source directions and physical speakers grows exponentially with time. We adopt a Bayesian framework and we propose a variational approximation of the posterior filtering distribution associated with multiple speaker tracking, as well as an efficient variational expectation-maximization (VEM) solver. The proposed online localization and tracking method is thoroughly evaluated using two datasets that contain recordings performed in real environments.Comment: IEEE Journal of Selected Topics in Signal Processing, 201

On binaural spatialization and the use of GPGPU for audio processing

3D recordings and audio, namely techniques that aim to create the perception of sound sources placed anywhere in 3 dimensional space, are becoming an interesting resource for composers, live performances and augmented reality. This thesis focuses on binaural spatialization techniques. We will tackle the problem from three different perspectives. The first one is related to the implementation of an engine for audio convolution, this is a real implementation problem where we will confront with a number of already available systems trying to achieve better results in terms of performances. General Purpose computing on Graphic Processing Units (GPGPU) is a promising approach to problems where a high parallelization of tasks is desirable. In this thesis the GPGPU approach is applied to both offline and real-time convolution having in mind the spatialization of multiple sound sources which is one of the critical problems in the field. Comparisons between this approach and typical CPU implementations are presented as well as between FFT and time domain approaches. The second aspect is related to the implementation of an augmented reality system having in mind an “off the shelf” system available to most home computers without the need of specialized hardware. A system capable of detecting the position of the listener through a head-tracking system and rendering a 3D audio environment by binaural spatialization is presented. Head tracking is performed through face tracking algorithms that use a standard webcam, and the result is presented over headphones, like in other typical binaural applications. With this system users can choose audio files to play, provide virtual positions for sources in an Euclidean space, and then listen as if they are coming from that position. If users move their head, the signals provided by the system change accordingly in real-time, thus providing the realistic effect of a coherent scene. The last aspect covered by this work is within the field of psychoacoustic, long term research where we are interested in understanding how binaural audio and recordings are perceived and how then auralization systems can be efficiently designed. Considerations with regard to the quality and the realism of such sounds in the context of ASA (Auditory Scene Analysis) are propose

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

International audienc

Auditory Displays and Assistive Technologies: the use of head movements by visually impaired individuals and their implementation in binaural interfaces

Visually impaired people rely upon audition for a variety of purposes, among these are the use of sound to identify the position of objects in their surrounding environment. This is limited not just to localising sound emitting objects, but also obstacles and environmental boundaries, thanks to their ability to extract information from reverberation and sound reflections- all of which can contribute to effective and safe navigation, as well as serving a function in certain assistive technologies thanks to the advent of binaural auditory virtual reality. It is known that head movements in the presence of sound elicit changes in the acoustical signals which arrive at each ear, and these changes can improve common auditory localisation problems in headphone-based auditory virtual reality, such as front-to-back reversals. The goal of the work presented here is to investigate whether the visually impaired naturally engage head movement to facilitate auditory perception and to what extent it may be applicable to the design of virtual auditory assistive technology. Three novel experiments are presented; a field study of head movement behaviour during navigation, a questionnaire assessing the self-reported use of head movement in auditory perception by visually impaired individuals (each comparing visually impaired and sighted participants) and an acoustical analysis of inter-aural differences and cross- correlations as a function of head angle and sound source distance. It is found that visually impaired people self-report using head movement for auditory distance perception. This is supported by head movements observed during the field study, whilst the acoustical analysis showed that interaural correlations for sound sources within 5m of the listener were reduced as head angle or distance to sound source were increased, and that interaural differences and correlations in reflected sound were generally lower than that of direct sound. Subsequently, relevant guidelines for designers of assistive auditory virtual reality are proposed