797 research outputs found
Didactic platform with a DSP to support the teaching of digital signal processing
Si ens posem en context d'un estudiant d'enginyeria, descobrirem que una de les majors motivacions de l'aprenentatge són les pràctiques de laboratori. Aquest treball de fi de grau tractarà sobre la recerca i el desenvolupament d'una plataforma didàctica per a l'assignatura de «Processament Digital del Senyal», impartida durant el tercer curs acadèmic del grau d'Enginyeria de Sistemes TIC. Aquesta plataforma que desenvoluparem inclourà un processador de senyals digitals (DSP) i els perifèrics necessaris perquè els estudiants i professors creïn projectes en un entorn de prototipatge ràpid. A més, els annexos proporcionats haurien de complir amb els requisits per a que aquells que estiguin interessats puguin fabricar el nostre disseny amb poques dificultats.If we put ourselves in the context of an engineering student, we will discover that one of the greatest motivations for learning are laboratory works. This final degree thesis will be about the research and development of a didactic platform for the subject of Digital Signal Processing, taught during the third academical year of the ICT Systems Engineering degree. This platform we are going to develop will encase a digital signal processor (DSP) and the required peripherals for the students and teachers to quickly create projects in a fast prototyping environment. Additionally, the provided annexes should meet with the requirements for those who are interested to manufacture our design with little trouble
Audio Engineering and Production at WPI
This project is the culmination of research and practice in the field of audio engineering. The aim is to propose a design for a new recording studio and audio engineering environment at WPI. This goal is achieved by researching numerous recording studios and performance venues in London during E Term, 2009 to decide on the best equipment and space for this environment. It is also achieved by completing professional audio production projects in the realms of multi-tracking and live ensemble, and establishing procedure for these kinds of projects in the future of WPI
Multi Channel Audio Environment
This project is dedicated to creating an interactive and immersive three-dimensional surround sound interface. Different from most 5.1 or 7.1 systems, this immersive environment combines sound wave mechanics with a 40 point source system to localize a sound in a user controlled 3D environment. An interactive and real-time system like this can be used to experiment with psychoacoustics, and to simulate complex environments where many sounds are coming from many different places
Design and Implementation of a Re-Configurable Arbitrary Signal Generator and Radio Frequency Spectrum Analyser
This research is focused on the design, simulation and implementation of a reconfigurable arbitrary signal generator and the design, simulation and implementation of a radio frequency spectrum analyser based on digital signal processing.
Until recently, Application Specific Integrated Circuits (ASICs) were used to produce high performance re-configurable function and arbitrary waveform generators with comprehensive modulation capabilities. However, that situation is now changing with the availability of advanced but low cost Field Programmable Gate Arrays (FPGAs), which could be used as an alternative to ASICs in these applications. The availability of high performance FPGA families opens up the opportunity to compete with ASIC solutions at a fraction of the development cost of an ASIC solution. A fast digital signal processing algorithm for digital waveform generation, using primarily but not limited to Direct Digital Synthesis (DDS) technologies, developed and implemented in a field-configurable logic, with control provided by an embedded microprocessor replacing a high cost ASIC design appeared to be a very attractive concept. This research demonstrates that such a concept is feasible in its entirety.
A fully functional, low-complexity, low cost, pulse, Gaussian white noise and DDS based function and arbitrary waveform generator, capable of being amplitude, frequency and phase modulated by an internally generated or external modulating signal was implemented in a low-cost FPGA. The FPGA also included the capabilities to perform pulse width modulation and pulse delay modulation on pulse waveforms. Algorithms to up-convert the sampling rate of the external modulating signal using Cascaded Integrator Comb (CIC) filters and using interpolation method were analysed. Both solutions were implemented to compare their hardware complexities. Analysis of generating noise with user-defined distribution is presented. The ability of triggering the generator by an internally generated or an external event to generate a burst of waveforms where the time between the trigger signal and waveform output is fixed was also implemented in the FPGA. Finally, design of interface to a microprocessor to provide control of the versatile waveform generator was also included in the FPGA. This thesis summarises the literature, design considerations, simulation and implementation of the generator design.
The second part of the research is focused on radio frequency spectrum analysis based on digital signal processing. Most existing spectrum analysers are analogue in nature and their complexity increases with frequency. Therefore, the possibility of using digital techniques for spectrum analysis was considered. The aim was to come up with digital system architecture for spectrum analysis and to develop and implement the new approach on a suitable digital platform.
This thesis analyses the current literature on shifting algorithms to remove spurious responses and highlights its drawbacks. This thesis also analyses existing literature on quadrature receivers and presents novel adaptation of the existing architectures for application in spectrum analysis. A wide band spectrum analyser receiver with compensation for gain and phase imbalances in the Radio Frequency (RF) input range, as well as compensation for gain and phase imbalances within the Intermediate Frequency (IF) pass band complete with Resolution Band Width (RBW) filtering, Video Band Width (VBW) filtering and amplitude detection was implemented in a low cost FPGA. The ability to extract the modulating signal from a frequency or amplitude modulated RF signal was also implemented. The same family of FPGA used in the generator design was chosen to be the digital platform for this design. This research makes arguments for the new architecture and then summarises the literature, design considerations, simulation and implementation of the new digital algorithm for the radio frequency spectrum analyser
High definition systems in Japan
The successful implementation of a strategy to produce high-definition systems within the Japanese economy will favorably affect the fundamental competitiveness of Japan relative to the rest of the world. The development of an infrastructure necessary to support high-definition products and systems in that country involves major commitments of engineering resources, plants and equipment, educational programs and funding. The results of these efforts appear to affect virtually every aspect of the Japanese industrial complex. The results of assessments of the current progress of Japan toward the development of high-definition products and systems are presented. The assessments are based on the findings of a panel of U.S. experts made up of individuals from U.S. academia and industry, and derived from a study of the Japanese literature combined with visits to the primary relevant industrial laboratories and development agencies in Japan. Specific coverage includes an evaluation of progress in R&D for high-definition television (HDTV) displays that are evolving in Japan; high-definition standards and equipment development; Japanese intentions for the use of HDTV; economic evaluation of Japan's public policy initiatives in support of high-definition systems; management analysis of Japan's strategy of leverage with respect to high-definition products and systems
Audio localization for mobile robots
The department of the University for which I worked is developing a project based on the interaction with robots in the environment. My work was to define an audio system for the robot. This audio system that I have to realize consists on a mobile head which is able to follow the sound in its environment. This subject was treated as a research problem, with the liberty to find and develop different solutions and make them evolve in the chosen way.Preprin
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
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