Digital Signal Processing Research Program

Contains table of contents for Section 2, an introduction, reports on sixteen research projects and a list of publications.Bose CorporationMIT-Woods Hole Oceanographic Institution Joint Graduate Program in Oceanographic EngineeringAdvanced Research Projects Agency/U.S. Navy - Office of Naval Research Grant N00014-93-1-0686Lockheed Sanders, Inc./U.S. Navy - Office of Naval Research Contract N00014-91-C-0125U.S. Air Force - Office of Scientific Research Grant AFOSR-91-0034AT&T Laboratories Doctoral Support ProgramAdvanced Research Projects Agency/U.S. Navy - Office of Naval Research Grant N00014-89-J-1489U.S. Navy - Office of Naval Research Grant N00014-93-1-0686National Science Foundation FellowshipMaryland Procurement Office Contract MDA904-93-C-4180U.S. Navy - Office of Naval Research Grant N00014-91-J-162

Interference Exploitation via Symbol-Level Precoding: Overview, State-of-the-Art and Future Directions

Interference is traditionally viewed as a performance limiting factor in wireless communication systems, which is to be minimized or mitigated. Nevertheless, a recent line of work has shown that by manipulating the interfering signals such that they add up constructively at the receiver side, known interference can be made beneficial and further improve the system performance in a variety of wireless scenarios, achieved by symbol-level precoding (SLP). This paper aims to provide a tutorial on interference exploitation techniques from the perspective of precoding design in a multi-antenna wireless communication system, by beginning with the classification of constructive interference (CI) and destructive interference (DI). The definition for CI is presented and the corresponding mathematical characterization is formulated for popular modulation types, based on which optimization-based precoding techniques are discussed. In addition, the extension of CI precoding to other application scenarios as well as for hardware efficiency is also described. Proof-of-concept testbeds are demonstrated for the potential practical implementation of CI precoding, and finally a list of open problems and practical challenges are presented to inspire and motivate further research directions in this area

Multimodal methods for blind source separation of audio sources

The enhancement of the performance of frequency domain convolutive blind source separation (FDCBSS) techniques when applied to the problem of separating audio sources recorded in a room environment is the focus of this thesis. This challenging application is termed the cocktail party problem and the ultimate aim would be to build a machine which matches the ability of a human being to solve this task. Human beings exploit both their eyes and their ears in solving this task and hence they adopt a multimodal approach, i.e. they exploit both audio and video modalities. New multimodal methods for blind source separation of audio sources are therefore proposed in this work as a step towards realizing such a machine. The geometry of the room environment is initially exploited to improve the separation performance of a FDCBSS algorithm. The positions of the human speakers are monitored by video cameras and this information is incorporated within the FDCBSS algorithm in the form of constraints added to the underlying cross-power spectral density matrix-based cost function which measures separation performance. [Continues.

COMPARISON METRICS AND PERFORMANCE ESTIMATIONS FOR DEEP BEAMFORMING DEEP NEURAL NETWORK BASED AUTOMATIC SPEECH RECOGNITION SYSTEMS USING MICROPHONE-ARRAYS

Automatic Speech Recognition (ASR) functionality, the automatic translation of speech into text, is on the rise today and is required for various use-cases, scenarios, and applications. An ASR engine by itself faces difficulties when encountering live input of audio data, regardless of how sophisticated and advanced it may be. That is especially true, under the circumstances such as a noisy ambient environment, multiple speakers, or faulty microphones. These kinds of challenges characterize a realistic scenario for an ASR system. ASR functionality continues to evolve toward more comprehensive End-to-End (E2E) solutions. E2E solution development focuses on three significant characteristics. The solution has to be robust enough to show endurance against external interferences. Also, it has to maintain flexibility, so it can easily extend in expectation of adapting to new scenarios or in order to achieve better performance. Lastly, we expect the solution to be modular enough to fit into new applications conveniently. Such an E2E ASR solution may include several additional micro-modules of speech enhancements besides the ASR engine, which is very complicated by itself. Adding these micro-modules can enhance the robustness and improve the overall system’s performance. Examples of such possible micro-modules include noise cancellation and speech separation, multi-microphone arrays, and adaptive beamformer(s). Being a comprehensive solution built of numerous micro-modules is technologically challenging to implement and challenging to integrate into resource-limited mobile systems. By offloading the complex computations to a server on the cloud, the system can fit more easily in less capable computing devices. Nevertheless, that compute offloading comes with the cost of giving up on real-time analysis, and increasing the overall system bandwidth. In addition, offloading to a server must have connectivity to the cloud over the internet. To find the optimal trade-offs between performance, Hardware (HW) and Software (SW) requirements or limitations, maximal computation time allowed for real-time analysis, and the detection accuracy, one should first define the different metrics used for the evaluation of such an E2E ASR system. Secondly, one needs to determine the extent of correlation between those metrics, plus the ability to forecast the impact each variation has on the others. This research presents novel progress in optimally designing a robust E2E-ASR system targeted for mobile, resource-limited devices. First, we describe evaluation metrics for each domain of interest, spread over vast engineering subjects. Here, we emphasize any bindings between the metrics across domains and the degree of impact derived from a change in the system’s specifications or constraints. Second, we present the effectiveness of applying machine learning techniques that can generalize and provide results of improved overall performance and robustness. Third, we present an approach of substituting architectures, changing algorithms, and approximating complex computations by utilizing a custom dedicated hardware acceleration in order to replace the traditional state-of-the-art SW-based solutions, thus providing real-time analysis capabilities to resource-limited systems

Developing a Noise-Robust Beat Learning Algorithm for Music-Information Retrieval

The field of Music-Information Retrieval (Music-IR) involves the development of algorithms that can analyze musical audio and extract various high-level musical features. Many such algorithms have been developed, and systems now exist that can reliably identify features such as beat locations, tempo, and rhythm from musical sources. These features in turn are used to assist in a variety of music-related tasks ranging from automatically creating playlists that match specified criteria to synchronizing various elements, such as computer graphics, with a performance. These Music-IR systems thus help humans to enjoy and interact with music. While current systems for identifying beats in music are have found widespread utility, most of them have been developed on music that is relatively free of acoustic noise. Much of the music that humans listen to, though, is performed in noisy environments. People often enjoy music in crowded clubs and noisy rooms, but this music is much more challenging for Music-IR systems to analyze, and current beat trackers generally perform poorly on musical audio heard in such conditions. If our algorithms could accurately process this music, though, it would enable this music too to be used in applications such as automatic song selection, which are currently limited to music taken directly from professionally-produced digital files that have little acoustic noise. Noise-robust beat learning algorithms would also allow for additional types of performance augmentation which create noise and thus cannot be used with current algorithms. Such a system, for instance, could aid robots in performing synchronously with music, whereas current systems are generally unable to accurately process audio heard in conjunction with noisy robot motors. This work aims to present a new approach for learning beats and identifying both their temporal locations and their spectral characteristics for music recorded in the presence of noise. First, datasets of musical audio recorded in environments with multiple types of noise were collected and annotated. Noise sources used for these datasets included HVAC sounds from a room, chatter from a crowded bar, and fans and motor noises from a moving robot. Second, an algorithm for learning and locating musical beats was developed which incorporates signal processing and machine learning techniques such as Harmonic-Percussive Source Separation and Probabilistic Latent Component Analysis. A representation of the musical signal called the stacked spectrogram was also utilized in order to better represent the time-varying nature of the beats. Unlike many current systems, which assume that the beat locations will be correlated with some hand-crafted features, this system learns the beats directly from the acoustic signal. Finally, the algorithm was tested against several state-of-the-art beat trackers on the audio datasets. The resultant system was found to significantly outperform the state-of-the-art when evaluated on audio played in realistically noisy conditions.Ph.D., Electrical Engineering -- Drexel University, 201

Single-Microphone Speech Enhancement and Separation Using Deep Learning

The cocktail party problem comprises the challenging task of understanding a speech signal in a complex acoustic environment, where multiple speakers and background noise signals simultaneously interfere with the speech signal of interest. A signal processing algorithm that can effectively increase the speech intelligibility and quality of speech signals in such complicated acoustic situations is highly desirable. Especially for applications involving mobile communication devices and hearing assistive devices. Due to the re-emergence of machine learning techniques, today, known as deep learning, the challenges involved with such algorithms might be overcome. In this PhD thesis, we study and develop deep learning-based techniques for two sub-disciplines of the cocktail party problem: single-microphone speech enhancement and single-microphone multi-talker speech separation. Specifically, we conduct in-depth empirical analysis of the generalizability capability of modern deep learning-based single-microphone speech enhancement algorithms. We show that performance of such algorithms is closely linked to the training data, and good generalizability can be achieved with carefully designed training data. Furthermore, we propose uPIT, a deep learning-based algorithm for single-microphone speech separation and we report state-of-the-art results on a speaker-independent multi-talker speech separation task. Additionally, we show that uPIT works well for joint speech separation and enhancement without explicit prior knowledge about the noise type or number of speakers. Finally, we show that deep learning-based speech enhancement algorithms designed to minimize the classical short-time spectral amplitude mean squared error leads to enhanced speech signals which are essentially optimal in terms of STOI, a state-of-the-art speech intelligibility estimator.Comment: PhD Thesis. 233 page

Aeronautical engineering: A continuing bibliography with indexes (supplement 257)

This bibliography lists 560 reports, articles, and other documents introduced into the NASA scientific and technical information system in September 1990. Subject coverage includes: design, construction and testing of aircraft and aircraft engines; aircraft components, equipment and systems; ground support systems; and theoretical and applied aspects of aerodynamics and general fluid dynamics