Speech Quality Evaluation based on AM-FM time-frequency representations

This paper deals with the application of information extracted from AM and FM time-frequency representations of speech to the task of determining speech quality. The representations are introduced and then the procedure for data extraction is outlined. The experimental setup for the assessment of objective quality covers distortions typically found in speech communication systems. To determine how well these quality measures perform, regression analysis is used to evaluate how well they estimate the results of subjective testing. Considering each class of distortions individually the objective measures demonstrate good performance, however, this level does not seem to hold as well in the aggregate case. This leads to suggestions as to where possible improvements can be made to the procedure

SwG-former: Sliding-window Graph Convolutional Network Integrated with Conformer for Sound Event Localization and Detection

Sound event localization and detection (SELD) is a joint task of sound event detection (SED) and direction of arrival (DoA) estimation. SED mainly relies on temporal dependencies to distinguish different sound classes, while DoA estimation depends on spatial correlations to estimate source directions. To jointly optimize two subtasks, the SELD system should extract spatial correlations and model temporal dependencies simultaneously. However, numerous models mainly extract spatial correlations and model temporal dependencies separately. In this paper, the interdependence of spatial-temporal information in audio signals is exploited for simultaneous extraction to enhance the model performance. In response, a novel graph representation leveraging graph convolutional network (GCN) in non-Euclidean space is developed to extract spatial-temporal information concurrently. A sliding-window graph (SwG) module is designed based on the graph representation. It exploits sliding-windows with different sizes to learn temporal context information and dynamically constructs graph vertices in the frequency-channel (F-C) domain to capture spatial correlations. Furthermore, as the cornerstone of message passing, a robust Conv2dAgg function is proposed and embedded into the SwG module to aggregate the features of neighbor vertices. To improve the performance of SELD in a natural spatial acoustic environment, a general and efficient SwG-former model is proposed by integrating the SwG module with the Conformer. It exhibits superior performance in comparison to recent advanced SELD models. To further validate the generality and efficiency of the SwG-former, it is seamlessly integrated into the event-independent network version 2 (EINV2) called SwG-EINV2. The SwG-EINV2 surpasses the state-of-the-art (SOTA) methods under the same acoustic environment

Audio Processing and Loudness Estimation Algorithms with iOS Simulations

abstract: The processing power and storage capacity of portable devices have improved considerably over the past decade. This has motivated the implementation of sophisticated audio and other signal processing algorithms on such mobile devices. Of particular interest in this thesis is audio/speech processing based on perceptual criteria. Specifically, estimation of parameters from human auditory models, such as auditory patterns and loudness, involves computationally intensive operations which can strain device resources. Hence, strategies for implementing computationally efficient human auditory models for loudness estimation have been studied in this thesis. Existing algorithms for reducing computations in auditory pattern and loudness estimation have been examined and improved algorithms have been proposed to overcome limitations of these methods. In addition, real-time applications such as perceptual loudness estimation and loudness equalization using auditory models have also been implemented. A software implementation of loudness estimation on iOS devices is also reported in this thesis. In addition to the loudness estimation algorithms and software, in this thesis project we also created new illustrations of speech and audio processing concepts for research and education. As a result, a new suite of speech/audio DSP functions was developed and integrated as part of the award-winning educational iOS App 'iJDSP." These functions are described in detail in this thesis. Several enhancements in the architecture of the application have also been introduced for providing the supporting framework for speech/audio processing. Frame-by-frame processing and visualization functionalities have been developed to facilitate speech/audio processing. In addition, facilities for easy sound recording, processing and audio rendering have also been developed to provide students, practitioners and researchers with an enriched DSP simulation tool. Simulations and assessments have been also developed for use in classes and training of practitioners and students.Dissertation/ThesisM.S. Electrical Engineering 201

Space/time/frequency methods in adaptive radar

Radar systems may be processed with various space, time and frequency techniques. Advanced radar systems are required to detect targets in the presence of jamming and clutter. This work studies the application of two types of radar systems. It is well known that targets moving along-track within a Synthetic Aperture Radar field of view are imaged as defocused objects. The SAR stripmap mode is tuned to stationary ground targets and the mismatch between the SAR processing parameters and the target motion parameters causes the energy to spill over to adjacent image pixels, thus hindering target feature extraction and reducing the probability of detection. The problem can be remedied by generating the image using a filter matched to the actual target motion parameters, effectively focusing the SAR image on the target. For a fixed rate of motion the target velocity can be estimated from the slope of the Doppler frequency characteristic. The problem is similar to the classical problem of estimating the instantaneous frequency of a linear FM signal (chirp). The Wigner-Ville distribution, the Gabor expansion, the Short-Time Fourier transform and the Continuous Wavelet Transform are compared with respect to their performance in noisy SAR data to estimate the instantaneous Doppler frequency of range compressed SAR data. It is shown that these methods exhibit sharp signal-to-noise threshold effects. The space-time radar problem is well suited to the application of techniques that take advantage of the low-rank property of the space-time covariance matrix. It is shown that reduced-rank methods outperform full-rank space-time adaptive processing when the space-time covariance matrix is estimated from a dataset with limited support. The utility of reduced-rank methods is demonstrated by theoretical analysis, simulations and analysis of real data. It is shown that reduced-rank processing has two effects on the performance: increased statistical stability which tends to improve performance, and introduction of a bias which lowers the signal-to-noise ratio. A method for evaluating the theoretical conditioned SNR for fixed reduced-rank transforms is also presented

Variability of Wind and Wind Power

Non

BeliefPPG: Uncertainty-aware Heart Rate Estimation from PPG signals via Belief Propagation

We present a novel learning-based method that achieves state-of-the-art performance on several heart rate estimation benchmarks extracted from photoplethysmography signals (PPG). We consider the evolution of the heart rate in the context of a discrete-time stochastic process that we represent as a hidden Markov model. We derive a distribution over possible heart rate values for a given PPG signal window through a trained neural network. Using belief propagation, we incorporate the statistical distribution of heart rate changes to refine these estimates in a temporal context. From this, we obtain a quantized probability distribution over the range of possible heart rate values that captures a meaningful and well-calibrated estimate of the inherent predictive uncertainty. We show the robustness of our method on eight public datasets with three different cross-validation experiments.Comment: Conference on Uncertainty in Artificial Intelligence (UAI) 2023. The first two authors contributed equall

Noise Characteristics of Operational Real‐Time High‐Rate GNSS Positions in a Large Aperture Network

Large earthquakes are difficult to model in real‐time with traditional inertial seismic measurements. Several algorithms that leverage high‐rate real‐time Global Navigation Satellite Systems (HR‐GNSS) positions have been proposed, and it has been shown that they can supplement the earthquake monitoring effort. However, analyses of the long‐term noise behavior of high‐rate real‐time GNSS positions, which are important to understand how the data can be used operationally by monitoring agencies, have been limited to just a few sites and to short time spans. Here, we show results from an analysis of the noise characteristics of 1 year of positions at 213 GNSS sites spanning a large geographic region from Southern California to Alaska. We characterize the behavior of noise and propose several references noise models which can be used as baselines to compare against as technological improvements allow for higher precision solutions. We also show how to use the reference noise models to generate realistic synthetic noise that can be used in simulations of HR‐GNSS waveforms. We discuss spatiotemporal variations in the noise and their potential sources and significance. We also detail how noise analysis can be used in a dynamic quality control to determine which sites should or should not contribute positions to an earthquake modeling algorithm at a particular moment in time. We posit that while there remain important improvements yet to be made, such as reducing the number of outliers in the time series, the present quality of real‐time HR‐GNSS waveforms is more than sufficient for monitoring large earthquakes

Non-Negative Matrix Factorization Based Algorithms to Cluster Frequency Basis Functions for Monaural Sound Source Separation.

Monophonic sound source separation (SSS) refers to a process that separates out audio signals produced from the individual sound sources in a given acoustic mixture, when the mixture signal is recorded using one microphone or is directly recorded onto one reproduction channel. Many audio applications such as pitch modification and automatic music transcription would benefit from the availability of segregated sound sources from the mixture of audio signals for further processing. Recently, Non-negative matrix factorization (NMF) has found application in monaural audio source separation due to its ability to factorize audio spectrograms into additive part-based basis functions, where the parts typically correspond to individual notes or chords in music. An advantage of NMF is that there can be a single basis function for each note played by a given instrument, thereby capturing changes in timbre with pitch for each instrument or source. However, these basis functions need to be clustered to their respective sources for the reconstruction of the individual source signals. Many clustering methods have been proposed to map the separated signals into sources with considerable success. Recently, to avoid the need of clustering, Shifted NMF (SNMF) was proposed, which assumes that the timbre of a note is constant for all the pitches produced by an instrument. SNMF has two drawbacks. Firstly, the assumption that the timbre of the notes played by an instrument remains constant, is not true in general. Secondly, the SNMF method uses the Constant Q transform (CQT) and the lack of a true inverse of the CQT results in compromising on separation quality of the reconstructed signal. The principal aim of this thesis is to attempt to solve the problem of clustering NMF basis functions. Our first major contribution is the use of SNMF as a method of clustering the basis functions obtained via standard NMF. The proposed SNMF clustering method aims to cluster the frequency basis functions obtained via standard NMF to their respective sources by making use of shift invariance in a log-frequency domain. Further, a minor contribution is made by improving the separation performance of the standard SNMF algorithm (here used directly to separate sources) obtained through the use of an improved inverse CQT. Here, the standard SNMF algorithm finds shift-invariance in a CQ spectrogram, that contain the frequency basis functions, obtained directly from the spectrogram of the audio mixture. Our next contribution is an improvement in the SNMF clustering algorithm through the incorporation of the CQT matrix inside the SNMF model in order to avoid the need of an inverse CQT to reconstruct the clustered NMF basis unctions. Another major contribution deals with the incorporation of a constraint called group sparsity (GS) into the SNMF clustering algorithm at two stages to improve clustering. The effect of the GS is evaluated on various SNMF clustering algorithms proposed in this thesis. Finally, we have introduced a new family of masks to reconstruct the original signal from the clustered basis functions and compared their performance to the generalized Wiener filter masks using three different factorisation-based separation algorithms. We show that better separation performance can be achieved by using the proposed family of masks

Trennung und Schätzung der Anzahl von Audiosignalquellen mit Zeit- und Frequenzüberlappung

Everyday audio recordings involve mixture signals: music contains a mixture of instruments; in a meeting or conference, there is a mixture of human voices. For these mixtures, automatically separating or estimating the number of sources is a challenging task. A common assumption when processing mixtures in the time-frequency domain is that sources are not fully overlapped. However, in this work we consider some cases where the overlap is severe — for instance, when instruments play the same note (unison) or when many people speak concurrently ("cocktail party") — highlighting the need for new representations and more powerful models. To address the problems of source separation and count estimation, we use conventional signal processing techniques as well as deep neural networks (DNN). We first address the source separation problem for unison instrument mixtures, studying the distinct spectro-temporal modulations caused by vibrato. To exploit these modulations, we developed a method based on time warping, informed by an estimate of the fundamental frequency. For cases where such estimates are not available, we present an unsupervised model, inspired by the way humans group time-varying sources (common fate). This contribution comes with a novel representation that improves separation for overlapped and modulated sources on unison mixtures but also improves vocal and accompaniment separation when used as an input for a DNN model. Then, we focus on estimating the number of sources in a mixture, which is important for real-world scenarios. Our work on count estimation was motivated by a study on how humans can address this task, which lead us to conduct listening experiments, confirming that humans are only able to estimate the number of up to four sources correctly. To answer the question of whether machines can perform similarly, we present a DNN architecture, trained to estimate the number of concurrent speakers. Our results show improvements compared to other methods, and the model even outperformed humans on the same task. In both the source separation and source count estimation tasks, the key contribution of this thesis is the concept of “modulation”, which is important to computationally mimic human performance. Our proposed Common Fate Transform is an adequate representation to disentangle overlapping signals for separation, and an inspection of our DNN count estimation model revealed that it proceeds to find modulation-like intermediate features.Im Alltag sind wir von gemischten Signalen umgeben: Musik besteht aus einer Mischung von Instrumenten; in einem Meeting oder auf einer Konferenz sind wir einer Mischung menschlicher Stimmen ausgesetzt. Für diese Mischungen ist die automatische Quellentrennung oder die Bestimmung der Anzahl an Quellen eine anspruchsvolle Aufgabe. Eine häufige Annahme bei der Verarbeitung von gemischten Signalen im Zeit-Frequenzbereich ist, dass die Quellen sich nicht vollständig überlappen. In dieser Arbeit betrachten wir jedoch einige Fälle, in denen die Überlappung immens ist zum Beispiel, wenn Instrumente den gleichen Ton spielen (unisono) oder wenn viele Menschen gleichzeitig sprechen (Cocktailparty) —, so dass neue Signal-Repräsentationen und leistungsfähigere Modelle notwendig sind. Um die zwei genannten Probleme zu bewältigen, verwenden wir sowohl konventionelle Signalverbeitungsmethoden als auch tiefgehende neuronale Netze (DNN). Wir gehen zunächst auf das Problem der Quellentrennung für Unisono-Instrumentenmischungen ein und untersuchen die speziellen, durch Vibrato ausgelösten, zeitlich-spektralen Modulationen. Um diese Modulationen auszunutzen entwickelten wir eine Methode, die auf Zeitverzerrung basiert und eine Schätzung der Grundfrequenz als zusätzliche Information nutzt. Für Fälle, in denen diese Schätzungen nicht verfügbar sind, stellen wir ein unüberwachtes Modell vor, das inspiriert ist von der Art und Weise, wie Menschen zeitveränderliche Quellen gruppieren (Common Fate). Dieser Beitrag enthält eine neuartige Repräsentation, die die Separierbarkeit für überlappte und modulierte Quellen in Unisono-Mischungen erhöht, aber auch die Trennung in Gesang und Begleitung verbessert, wenn sie in einem DNN-Modell verwendet wird. Im Weiteren beschäftigen wir uns mit der Schätzung der Anzahl von Quellen in einer Mischung, was für reale Szenarien wichtig ist. Unsere Arbeit an der Schätzung der Anzahl war motiviert durch eine Studie, die zeigt, wie wir Menschen diese Aufgabe angehen. Dies hat uns dazu veranlasst, eigene Hörexperimente durchzuführen, die bestätigten, dass Menschen nur in der Lage sind, die Anzahl von bis zu vier Quellen korrekt abzuschätzen. Um nun die Frage zu beantworten, ob Maschinen dies ähnlich gut können, stellen wir eine DNN-Architektur vor, die erlernt hat, die Anzahl der gleichzeitig sprechenden Sprecher zu ermitteln. Die Ergebnisse zeigen Verbesserungen im Vergleich zu anderen Methoden, aber vor allem auch im Vergleich zu menschlichen Hörern. Sowohl bei der Quellentrennung als auch bei der Schätzung der Anzahl an Quellen ist ein Kernbeitrag dieser Arbeit das Konzept der “Modulation”, welches wichtig ist, um die Strategien von Menschen mittels Computern nachzuahmen. Unsere vorgeschlagene Common Fate Transformation ist eine adäquate Darstellung, um die Überlappung von Signalen für die Trennung zugänglich zu machen und eine Inspektion unseres DNN-Zählmodells ergab schließlich, dass sich auch hier modulationsähnliche Merkmale finden lassen