Sub-Nyquist Sampling: Bridging Theory and Practice

Sampling theory encompasses all aspects related to the conversion of continuous-time signals to discrete streams of numbers. The famous Shannon-Nyquist theorem has become a landmark in the development of digital signal processing. In modern applications, an increasingly number of functions is being pushed forward to sophisticated software algorithms, leaving only those delicate finely-tuned tasks for the circuit level. In this paper, we review sampling strategies which target reduction of the ADC rate below Nyquist. Our survey covers classic works from the early 50's of the previous century through recent publications from the past several years. The prime focus is bridging theory and practice, that is to pinpoint the potential of sub-Nyquist strategies to emerge from the math to the hardware. In that spirit, we integrate contemporary theoretical viewpoints, which study signal modeling in a union of subspaces, together with a taste of practical aspects, namely how the avant-garde modalities boil down to concrete signal processing systems. Our hope is that this presentation style will attract the interest of both researchers and engineers in the hope of promoting the sub-Nyquist premise into practical applications, and encouraging further research into this exciting new frontier.Comment: 48 pages, 18 figures, to appear in IEEE Signal Processing Magazin

Radio Astronomy

Contains research objectives, summary of research and reports on five research projects.National Aeronautics and Space Administration (Grant NGL 22-009-016)National Aeronautics and Space Administration (Grant NGR 22-009-421)Langley Research Center Contract NASI-10693National Science Foundation (Grants GP-20769)National Science Foundation (Grants GP-21348)National Science Foundation (Grants GP-14589)California Institute of Technology Contract 952568Sloan Fund for Basic Research (M.I.T., Grant 241

Design and testing of a 96-channel neural interface module for the Networked Neuroprosthesis system

Abstract Background The loss of motor functions resulting from spinal cord injury can have devastating implications on the quality of one’s life. Functional electrical stimulation has been used to help restore mobility, however, current functional electrical stimulation (FES) systems require residual movements to control stimulation patterns, which may be unintuitive and not useful for individuals with higher level cervical injuries. Brain machine interfaces (BMI) offer a promising approach for controlling such systems; however, they currently still require transcutaneous leads connecting indwelling electrodes to external recording devices. While several wireless BMI systems have been designed, high signal bandwidth requirements limit clinical translation. Case Western Reserve University has developed an implantable, modular FES system, the Networked Neuroprosthesis (NNP), to perform combinations of myoelectric recording and neural stimulation for controlling motor functions. However, currently the existing module capabilities are not sufficient for intracortical recordings. Methods Here we designed and tested a 1 × 4 cm, 96-channel neural recording module prototype to fit within the specifications to mate with the NNP. The neural recording module extracts power between 0.3–1 kHz, instead of transmitting the raw, high bandwidth neural data to decrease power requirements. Results The module consumed 33.6 mW while sampling 96 channels at approximately 2 kSps. We also investigated the relationship between average spiking band power and neural spike rate, which produced a maximum correlation of R = 0.8656 (Monkey N) and R = 0.8023 (Monkey W). Conclusion Our experimental results show that we can record and transmit 96 channels at 2ksps within the power restrictions of the NNP system and successfully communicate over the NNP network. We believe this device can be used as an extension to the NNP to produce a clinically viable, fully implantable, intracortically-controlled FES system and advance the field of bioelectronic medicine.https://deepblue.lib.umich.edu/bitstream/2027.42/147921/1/42234_2019_Article_19.pd

Biorthogonality in lapped transforms : a study in high-quality audio compression

Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1996.Includes bibliographical references (leaves 76-82).by Shiufun Cheung.Ph.D

SETI science working group report

This report covers the initial activities and deliberations of a continuing working group asked to assist the SETI Program Office at NASA. Seven chapters present the group's consensus on objectives, strategies, and plans for instrumental R&D and for a microwave search for extraterrestrial in intelligence (SETI) projected for the end of this decade. Thirteen appendixes reflect the views of their individual authors. Included are discussions of the 8-million-channel spectrum analyzer architecture and the proof-of-concept device under development; signal detection, recognition, and identification on-line in the presence of noise and radio interference; the 1-10 GHz sky survey and the 1-3 GHz targeted search envisaged; and the mutual interests of SETI and radio astronomy. The report ends with a selective, annotated SETI reading list of pro and contra SETI publications

Epälineaarisen signaaliriippuvan akustisen keilanmuodostajan reaaliaikaimplementaatio

A real-time acoustical beamforming system incorporating the cross pattern coherence (CroPaC) post filtering method is implemented in this thesis. The real-time implementation consists of a signal-independent beamformer that is used for spatial discrimination of a sound field. The signal of the beamformer is post filtered by modulating it with a parameter that is derived from the cross-spectrum of two directional microphone signals. The post filter is implemented to enhance performance of beamforming (increase in signal-to-noise ratio), because beamformers are not efficient in environments with high level of reverberation. The post filtering method has been previously implemented in MATLAB for non-real-time use, and this system is the first real-time implementation of an acoustical beamforming system utilizing it. The implementation is programmed in the programming language C for the graphical signal processing program Max developed by Cycling '74. It utilizes a time-frequency domain processing, and the spherical Fourier transform for a decomposition of a sound field into spherical harmonic signals. The implementation can be used with microphone arrays with maximum of 32 microphone capsules, which are laid over rigid sphere with uniform or nearly-uniform arrangements. The real-time implementation can be utilized in many applications, which require algorithm to work in real-time, such as teleconferencing and acoustical cameras.Tässä diplomityössä implementoidaan reaaliaikainen akustinen keilanmuodostusjärjestelmä signaalien väliseen koherenssiin perustuvalla (CroPaC) jälkisuodatuksella. Reaaliaikaimplementaatio koostuu signaaliriippumattomasta keilanmuodostajasta, jota käytetään äänikentän spatiaaliseen suodatukseen. Keilanmuodostajan signaalia jälkisuodatetaan moduloimalla sitä parametrilla, joka johdetaan kahden suuntamikrofonin signaalin välisestä koherenssista. Jälkisuodatus implementoidaan keilanmuodostajan suorituskyvyn parantamiseksi (signaali-kohina-suhteen kasvu), sillä keilanmuodostajat eivät ole tehokkaita kaiuntaisissa ympäristöissä. Jälkisuodatusmetodi on aikaisemmin implementoitu MATLABissa ei-reaaliaikakäyttöä varten. Tämän työn implementaatio on ensimmäinen reaaliaikainen akustinen keilanmuodostusjärjestelmä, joka hyödyntää CroPaC-jälkisuodatusta. Implementaatio on ohjelmoitu C-ohjelmointikielellä graafiselle signaalinprosessointityökalulle Max, jonka on kehittänyt Cycling '74. Prosessointi tapahtuu aika-taajuustasossa ja siinä hyödynnetään äänikentän dekompositiota palloharmonisiin signaaleihin. Implementaatiota voidaan käyttää mikrofoniryhmällä, jossa on korkeintaan 32 mikrofonikapselia, jotka on asetettu jäykän pallon päälle tasavälein tai lähes tasavälein. Reaaliaikaimplementaatiota voidaan hyödyntää lukuisissa sovelluksissa, jotka edellyttävät algoritmin reaaliaikaista toimintaa, esimerkiksi puhelinkokouksissa ja akustisissa kameroissa

Channelization for Multi-Standard Software-Defined Radio Base Stations

As the number of radio standards increase and spectrum resources come under more pressure, it becomes ever less efficient to reserve bands of spectrum for exclusive use by a single radio standard. Therefore, this work focuses on channelization structures compatible with spectrum sharing among multiple wireless standards and dynamic spectrum allocation in particular. A channelizer extracts independent communication channels from a wideband signal, and is one of the most computationally expensive components in a communications receiver. This work specifically focuses on non-uniform channelizers suitable for multi-standard Software-Defined Radio (SDR) base stations in general and public mobile radio base stations in particular. A comprehensive evaluation of non-uniform channelizers (existing and developed during the course of this work) shows that parallel and recombined variants of the Generalised Discrete Fourier Transform Modulated Filter Bank (GDFT-FB) represent the best trade-off between computational load and flexibility for dynamic spectrum allocation. Nevertheless, for base station applications (with many channels) very high filter orders may be required, making the channelizers difficult to physically implement. To mitigate this problem, multi-stage filtering techniques are applied to the GDFT-FB. It is shown that these multi-stage designs can significantly reduce the filter orders and number of operations required by the GDFT-FB. An alternative approach, applying frequency response masking techniques to the GDFT-FB prototype filter design, leads to even bigger reductions in the number of coefficients, but computational load is only reduced for oversampled configurations and then not as much as for the multi-stage designs. Both techniques render the implementation of GDFT-FB based non-uniform channelizers more practical. Finally, channelization solutions for some real-world spectrum sharing use cases are developed before some final physical implementation issues are considered

Signal Subspace Processing in the Beam Space of a True Time Delay Beamformer Bank

A number of techniques for Radio Frequency (RF) source location for wide bandwidth signals have been described that utilize coherent signal subspace processing, but often suffer from limitations such as the requirement for preliminary source location estimation, the need to apply the technique iteratively, computational expense or others. This dissertation examines a method that performs subspace processing of the data from a bank of true time delay beamformers. The spatial diversity of the beamformer bank alleviates the need for a preliminary estimate while simultaneously reducing the dimensionality of subsequent signal subspace processing resulting in computational efficiency. The pointing direction of the true time delay beams is independent of frequency, which results in a mapping from element space to beam space that is wide bandwidth in nature. This dissertation reviews previous methods, introduces the present method, presents simulation results that demonstrate the assertions, discusses an analysis of performance in relation to the Cramer-Rao Lower Bound (CRLB) with various levels of noise in the system, and discusses computational efficiency. One limitation of the method is that in practice it may be appropriate for systems that can tolerate a limited field of view. The application of Electronic Intelligence is one such application. This application is discussed as one that is appropriate for a method exhibiting high resolution of very wide bandwidth closely spaced sources and often does not require a wide field of view. In relation to system applications, this dissertation also discusses practical employment of the novel method in terms of antenna elements, arrays, platforms, engagement geometries, and other parameters. The true time delay beam space method is shown through modeling and simulation to be capable of resolving closely spaced very wideband sources over a relevant field of view in a single algorithmic pass, requiring no course preliminary estimation, and exhibiting low computational expense superior to many previous wideband coherent integration techniques