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A microcomputer implementation of real time, continuously programmable digital filters
Title
A microcomputer implementation of real time, continuously programmable digital filters.
Creator
Storma, William Edward, Simmons, Fred O., Engineering
Abstract / Description
University of Central Florida College of Engineering Thesis; When a filter transfer function in s is replaced with the bilinear transform in z, t he resulting discrete model represents the original continuous model within a second order accuracy of integration. A unique set of recently discovered minimum memory algorithms that perform the bilinear transform on a continuous transfer function are implemented on an INTEL 8080 microprocessor system. Scal1ng techniques are used to frequency scale...
Show moreUniversity of Central Florida College of Engineering Thesis; When a filter transfer function in s is replaced with the bilinear transform in z, t he resulting discrete model represents the original continuous model within a second order accuracy of integration. A unique set of recently discovered minimum memory algorithms that perform the bilinear transform on a continuous transfer function are implemented on an INTEL 8080 microprocessor system. Scal1ng techniques are used to frequency scale all transfer functions to a standardized frequency. All data words are represented in a signed binary double precision format to maintain higher calculation speed and accuracy. Three test case transfer functions of different order are implemented using the bilinear transform algorithms. First, the algorithms are used to generate the three discrete models. Second, the continuous time models are driven by a step input function, generating a continuous time output. Third, the step function input is discretized and used to drive the bilinear algorithm derived models. Finally, the discrete outputs are compared with the continuous time outputs to validate and evaluate the software techniques used to implement the bilinear algorithms, which imply that the techniques provide a basis for new hardware designs.
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Date Issued
1979
Identifier
CFR0003497, ucf:53140
Format
Document (PDF)
PURL
http://purl.flvc.org/ucf/fd/CFR0003497
DIGITAL SIGNAL PROCESSING TECHNIQUES FOR COHERENT OPTICAL COMMUNICATION
Title
DIGITAL SIGNAL PROCESSING TECHNIQUES FOR COHERENT OPTICAL COMMUNICATION.
Creator
Goldfarb, Gilad, Li, Guifang, University of Central Florida
Abstract / Description
Coherent detection with subsequent digital signal processing (DSP) is developed, analyzed theoretically and numerically and experimentally demonstrated in various fiber‐optic transmission scenarios. The use of DSP in conjunction with coherent detection unleashes the benefits of coherent detection which rely on the preservation of full information of the incoming field. These benefits include high receiver sensitivity, the ability to achieve high spectral‐efficiency and the use of...
Show moreCoherent detection with subsequent digital signal processing (DSP) is developed, analyzed theoretically and numerically and experimentally demonstrated in various fiber‐optic transmission scenarios. The use of DSP in conjunction with coherent detection unleashes the benefits of coherent detection which rely on the preservation of full information of the incoming field. These benefits include high receiver sensitivity, the ability to achieve high spectral‐efficiency and the use of advanced modulation formats. With the immense advancements in DSP speeds, many of the problems hindering the use of coherent detection in optical transmission systems have been eliminated. Most notably, DSP alleviates the need for hardware phase‐locking and polarization tracking, which can now be achieved in the digital domain. The complexity previously associated with coherent detection is hence significantly diminished and coherent detection is once again considered a feasible detection alternative. In this thesis, several aspects of coherent detection (with or without subsequent DSP) are addressed. Coherent detection is presented as a means to extend the dispersion limit of a duobinary signal using an analog decision‐directed phase‐lock loop. Analytical bit‐error ratio estimation for quadrature phase‐shift keying signals is derived. To validate the promise for high spectral efficiency, the orthogonal‐wavelength‐division multiplexing scheme is suggested. In this scheme the WDM channels are spaced at the symbol rate, thus achieving the spectral efficiency limit. Theory, simulation and experimental results demonstrate the feasibility of this approach. Infinite impulse response filtering is shown to be an efficient alternative to finite impulse response filtering for chromatic dispersion compensation. Theory, design considerations, simulation and experimental results relating to this topic are presented. Interaction between fiber dispersion and nonlinearity remains the last major challenge deterministic effects pose for long‐haul optical data transmission. Experimental results which demonstrate the possibility to digitally mitigate both dispersion and nonlinearity are presented. Impairment compensation is achieved using backward propagation by implementing the split‐step method. Efficient realizations of the dispersion compensation operator used in this implementation are considered. Infinite‐impulse response and wavelet‐based filtering are both investigated as a means to reduce the required computational load associated with signal backward‐propagation. Possible future research directions conclude this dissertation.
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Date Issued
2008
Identifier
CFE0002384, ucf:47763
Format
Document (PDF)
PURL
http://purl.flvc.org/ucf/fd/CFE0002384
Mode-Division Multiplexed Transmission in Few-mode Fibers
Title
Mode-Division Multiplexed Transmission in Few-mode Fibers.
Creator
Bai, Neng, Li, Guifang, Christodoulides, Demetrios, Schulzgen, Axel, Abouraddy, Ayman, Phillips, Ronald, Ip, Ezra, University of Central Florida
Abstract / Description
As a promising candidate to break the single-mode fiber capacity limit, mode-division multiplexing (MDM) explores the spatial dimension to increase transmission capacity in fiber-optic communication. Two linear impairments, namely loss and multimode interference, present fundamental challenges to implementing MDM. In this dissertation, techniques to resolve these two issues are presented.To de-multiplex signals subject to multimode interference in MDM, Multiple-Input-Multiple-Output (MIMO)...
Show moreAs a promising candidate to break the single-mode fiber capacity limit, mode-division multiplexing (MDM) explores the spatial dimension to increase transmission capacity in fiber-optic communication. Two linear impairments, namely loss and multimode interference, present fundamental challenges to implementing MDM. In this dissertation, techniques to resolve these two issues are presented.To de-multiplex signals subject to multimode interference in MDM, Multiple-Input-Multiple-Output (MIMO) processing using adaptive frequency-domain equalization (FDE) is proposed and investigated. Both simulations and experiments validate that FDE can reduce the algorithmic complexity significantly in comparison with the conventional time-domain equalization (TDE) while achieving similar performance as TDE. To further improve the performance of FDE, two modifications on traditional FDE algorithm are demonstrated. i) normalized adaptive FDE is applied to increase the convergence speed by 5 times; ii) master-slave carrier recovery is proposed to reduce the algorithmic complexity of phase estimation by number of modes.Although FDE can reduce the computational complexity of the MIMO processing, due to large mode group delay (MGD) of FMF link and block processing, the algorithm still requires enormous memory and high hardware complexity. In order to reduce the required tap length (RTL) of the equalizer, differential mode group delay compensated fiber (DMGDC) has been proposed. In this dissertation, the analytical expression for RTL is derived for DMGDC systems under the weak mode coupling assumption. Instead of depending on the overall MGD of the link in DMGD uncompensated (DMGDUC) systems, the RTL of DMGDC systems depend on the MGD of a single DMGDC fiber section. The theoretical and numerical results suggest that by using small compensation step-size, the RTL of DMGDC link can be reduced by 2 orders of magnitude compared to DMGDUC link. To compensate the loss of different modes, multimode EDFAs are presented with re-configurable multimode pumps. By tuning the mode content of the multimode pump, mode-dependent gain (MDG) can be controlled and equalized. A proto-type FM-EDFA which could support 2 LP modes was constructed. The experimental results show that by using high order mode pumps, the modal gain difference can be reduced. By applying both multimode EDFA and equalization techniques, 26.4Tb/s MDM-WDM transmission was successfully demonstrated.A brief summary and several possible future research directions conclude this dissertation.
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Date Issued
2013
Identifier
CFE0004811, ucf:49751
Format
Document (PDF)
PURL
http://purl.flvc.org/ucf/fd/CFE0004811
DISCUSSION ON EFFECTIVE RESTORATION OF ORAL SPEECH USING VOICE CONVERSION TECHNIQUES BASED ON GAUSSIAN MIXTURE MODELING
Title
DISCUSSION ON EFFECTIVE RESTORATION OF ORAL SPEECH USING VOICE CONVERSION TECHNIQUES BASED ON GAUSSIAN MIXTURE MODELING.
Creator
Alverio, Gustavo, Mikhael, Wasfy, University of Central Florida
Abstract / Description
Today's world consists of many ways to communicate information. One of the most effective ways to communicate is through the use of speech. Unfortunately many lose the ability to converse. This in turn leads to a large negative psychological impact. In addition, skills such as lecturing and singing must now be restored via other methods. The usage of text-to-speech synthesis has been a popular resolution of restoring the capability to use oral speech. Text to speech synthesizers convert...
Show moreToday's world consists of many ways to communicate information. One of the most effective ways to communicate is through the use of speech. Unfortunately many lose the ability to converse. This in turn leads to a large negative psychological impact. In addition, skills such as lecturing and singing must now be restored via other methods. The usage of text-to-speech synthesis has been a popular resolution of restoring the capability to use oral speech. Text to speech synthesizers convert text into speech. Although text to speech systems are useful, they only allow for few default voice selections that do not represent that of the user. In order to achieve total restoration, voice conversion must be introduced. Voice conversion is a method that adjusts a source voice to sound like a target voice. Voice conversion consists of a training and converting process. The training process is conducted by composing a speech corpus to be spoken by both source and target voice. The speech corpus should encompass a variety of speech sounds. Once training is finished, the conversion function is employed to transform the source voice into the target voice. Effectively, voice conversion allows for a speaker to sound like any other person. Therefore, voice conversion can be applied to alter the voice output of a text to speech system to produce the target voice. The thesis investigates how one approach, specifically the usage of voice conversion using Gaussian mixture modeling, can be applied to alter the voice output of a text to speech synthesis system. Researchers found that acceptable results can be obtained from using these methods. Although voice conversion and text to speech synthesis are effective in restoring voice, a sample of the speaker before voice loss must be used during the training process. Therefore it is vital that voice samples are made to combat voice loss.
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Date Issued
2007
Identifier
CFE0001793, ucf:47286
Format
Document (PDF)
PURL
http://purl.flvc.org/ucf/fd/CFE0001793