International Telemetering Conference Proceedings / October 25-28, 1999 / Riviera Hotel and Convention Center, Las Vegas, Nevada / During the cold war, undersea warfare (USW) was perceived as a large-area, deep-water operation. The need for USW has recently shifted to the shallow water, littoral environment. Consequently, US naval forces must train to operate in these littoral environments where regional conflicts are most likely to occur. In light of these requirements the Shallow Water Training Range (SWTR) has been initiated. Telemetry is used in the following areas of SWTR: fiber optic, microwave, RF and underwater. Only phase 1 of 8 phases of the program is executing therefore SWTR is a good opportunity for telemetry industry involvement.

International Telemetering Conference Proceedings / October 28-31, 1996 / Town and Country Hotel and Convention Center, San Diego, California / The monitoring of multi phase 400 Hz aircraft power includes monitoring the phase voltages, currents, real powers, and frequency. This paper describes the design of a multi channel card that uses digital signal processing (DSP) to measure these parameters on a cycle by cycle basis. The card measures the average, peak, minimum cycle, and maximum cycle values of these parameters.

International Telemetering Conference Proceedings / October 25-28, 1999 / Riviera Hotel and Convention Center, Las Vegas, Nevada / There are two misty understandings about PCM telemetry system in conventional concept: Waveform can not be restored accurately; to be restored accurately, a measured signal must be sampled at a higher sample rate. This paper discusses that by making use of multi-sample rate DSP technology, the sample rate of a measured signal can be reduced in transmission equipment, or system precision can be retained even if the performance of low pass filter declined.

Directly modulated lasers (DMLs) are a low cost solution for moderate reach systems due to their small footprint, low power dissipation and high output optical power. However, commercial 10-Gb/s on-off keying DMLs have been limited by an inherent modulation of the optical phase that accompanies the desired modulation of the optical intensity, which reduces their transmission distance to below 20 km. Furthermore, the ability to generate bit rates beyond 10-Gb/s with advanced modulation formats has been limited by the strict requirements on the laser drive current. The primary objective of this research is to dramatically enhance the capability of DML based systems through precise control over the drive current. This is achieved by digital signal processing (DSP) combined with a single digital-to-analog converter (DAC). In this research, a novel method to pre-compensate dispersion for metro and regional networks is demonstrated at 10.709-Gb/s. A look-up table (LUT) for the driving current is optimized for dispersion mitigation. Experimental results show a 25 fold increase in the transmission reach of a DML from 10 km to 252 km. A similar approach applied to a directly modulated chirp managed laser reveals a remarkable increase in the achievable transmission reach from 200 km to 608 km. In the context of access networks the DSP and DAC configuration is utilized for directly modulating a passive feedback laser (PFL) to generate differential phase shift keying (DPSK) signals at bit rates of 10.709-Gb/s, 14-Gb/s and 16-Gb/s. The quality of the DPSK signals is assessed using both noncoherent detection for a bit rate of 10.709-Gb/s and coherent detection with DSP involving a LUT pattern-dependent distortion compensator. For very short reach optical links, a 16-ary quadrature amplitude modulation signal is generated using subcarrier modulation with a subcarrier frequency of half the symbol rate, Nyquist pulse shaping, and a directly modulated PFL at bit rates up to 56-Gb/s. Using polarization multiplexing emulation, a pre-amplified direct detection receiver and DSP, loss margins of 12.6 dB and 8 dB are achieved for a 112-Gb/s dual polarization signal within a 33 GHz optical bandwidth at back-to-back and after 4 km transmission, respectively. / Thesis (Ph.D, Electrical & Computer Engineering) -- Queen's University, 2013-01-31 13:58:56.327

International Telemetering Conference Proceedings / October 30-November 02, 1995 / Riviera Hotel, Las Vegas, Nevada / In this paper, a new concept in PCM telemetry encoding equipment is described. Existing "programmable" PCM encoders allow only simple changes in the functionality of the hardware, such as input gain, offset, and word formatting. More importantly, these encoders do not provide capability for "in-flight" processing of signals and in general have not taken advantage of existing hardware and software digital signal processing technology. In-flight processing of signals can provide a significant reduction in the required transmission bandwidth, allowing additional data that may not have otherwise been transmitted to be sent on the telemetry channel. A modular digital signal processor (DSP) based PCM encoder architecture is described that has a set of on-board processing algorithms configurable via a simple-to-use graphical user interface. Algorithms included are compression (lossy and lossless), Fourier transforms of various resolutions (typically followed by peak detection to provide a data rate reduction), extreme values (max, min, rms), time filtering, regression, trajectory prediction, and serial data stream processing. Custom algorithms can be developed and included as part of the suite of processing algorithms. The preprocessing algorithms exist as firmware on the DSPs and can accommodate as many different signals as the processing bandwidth of the DSP can handle. Typically one DSP can handle many input signals and different algorithms. The encoder is programmable via a standard RS-232 serial interface allowing the signal input configuration, telemetry frame layout, and on-board processing algorithms to be changed quickly.

International Telemetering Conference Proceedings / October 26-29, 1992 / Town and Country Hotel and Convention Center, San Diego, California / This paper addresses a method to provide both secure and non-secure voice communications to a DS-1 network from a common keyset. In order to comply with both the electrical isolation requirements and the operational security issues regarding voice communications, an all-digital approach to the keyset was developed based upon the AD2101 DSP. Protocols that are handled by the keyset include: Multiple PTT modes, hot mike, telephone access, priority override, direct access, indirect access, paging, and monitor only. Special features that are addressed include: independent channel by channel assignment of access protocols, headset assignment, speaker assignment, and PTT assignment. Multiple microprocessors are used to implement the foregoing as well as down-loadable configurations, remote keyset control and monitoring, and composite audio outputs. Partitioning of the digital design provides RED to BLACK channel isolation and RED channel to AC power isolation of greater than 107 dB.

Despite the increasing sophistication of digital musical instruments, many performers, composer and listeners remain captivated by traditional acoustic instruments. Interest has grown in the past 2 decades in augmenting acoustic instruments with sensor and actuator technology and integrated digital signal processing, expanding the instrument's capabilities while retaining its essential acoustic character. In this thesis we present a technique, travelling wave control, which allows active control of the vibrations of musical strings and yet has been little explored in the musical instrument literature to date. The thesis seeks to demonstrate that travelling wave control is capable of active damping and of modifying the timbre of a musical string in ways that go beyond those available through the more conventional modal control paradigm. However, we show that travelling wave control is highly sensitive to nonlinearity, which in practical settings can lead to harmonic distortion and even instability in the string response. To avoid these problems, we design and build a highly linear optical string displacement sensor, and investigate the use of piezoelectric stacks to actuate the termination point of a string. With these components we design and build a functioning travelling wave control system which is capable of damping the vibrations of a plucked string without adversely affecting its timbre. We go on to show that by deliberately adding nonlinearity into the control system, we are able to modify the timbre of the string in a natural way by affecting the evolution of the modal amplitudes. The results demonstrate the feasibility of the concept and lay the groundwork for future integration of travelling wave control into future actuated musical instruments.

When a speaker is addressing an audience, a PA system consisting of a microphone and a loudspeaker is often used. If the microphone picks up too much of the loudspeaker energy, acoustic feedback in the form of an unwanted characteristic howling can occur. Limes Audio is a software company that specializes in improving sound quality in digital communications, mainly conference telephony, and has developed a reference product, the Magneto mixer, to demonstrate the capability of their software TrueVoice. The company now wishes to expand the field of usage for the Magneto mixer to enable it to work as a microphone mixer in PA scenarios, and for this, a feedback suppression feature is needed. This master’s thesis aims at surveying the market and the literature in the field and specifying the requirements for a feedback suppression feature. Three methods for suppressing howling feedback are evaluated through simulations and compared in terms of maximum stable gain (MSG) and subjective listening experience. The method that performed the best based on these criteria was acoustic feedback cancellation with a 5 Hz frequency shift on the loudspeaker signal. This method makes use of an adaptive filter to model the acoustic feedback path and to remove the feedback component from the microphone signal. In the simulations, the method was able to increase the stable gain by approximately 10 dB while maintaining a good sound quality. / När en talare talar för en publik används ofta ett PA system bestående av en mikrofon och en högtalare. Om mikrofonen tar upp för mycket av ljudet från högtalaren finns en överhängande risk för akustisk rundgång i form av ett karaktäristiskt oönskat tjut. Limes Audio är ett företag som utvecklar mjukvara för att förbättra ljudkvaliten i digital kommunikation, främst inom konferenstelefoni. De har utvecklat en demonstrationsprodukt, Magnetomixern, som kan användas som en konferenstelefon för att demonstrera deras programvara TrueVoice. Företaget önskar nu utveckla Magnetomixern till att även fungera som en ljudmixer för PA-scenarion, eller konferenstelefoni där intern ljudförstärkning i rummet behövs, och för detta behövs en funktion för att ta bort eventuell rundgång. Detta examensarbete har som mål att lägga grunden för en sådan funktion i Magnetomixern genom att undersöka marknaden och litteraturen på området. Tre metoder för att eliminera rundgång utvärderas i simuleringar och jämförs beträffande maximal stabil förstärkning (MSG) och subjektiv ljudkvalitet. Metoden ”Acoustic feedback cancellation” tillsammans med ett 5 Hz frekvensskifte på högtalarsignalen gav högst MSG och bäst ljudkvalitet. Metoden använder ett adaptivt filter för att approximera den akustiska återkopplingsvägen mellan högtalare och mikrofon samt tar bort rundgångskomponenter från mikrofonsignalen. I simuleringarna kunde metoden öka den maximala stabila förstärkningen med upp till 10 dB medan en god ljudkvalitet på talet bibehölls.

This pro ject summarizes the design and implementation of field programmable gate array (FPGA) based digital signal processing (DSP) hardware meant to be used in a software radio system. The filters and processing were first designed in MATLAB and then implemented using very high speed integrated circuit hardware description language (VHDL). Since this hardware is meant for a software radio system, making the hardware flexible was the main design goal. Flexibility in the FPGA design was reached using VHDL generics and generate for loops. The hardware was verified using MATLAB generated signals as stimulus to the VHDL design and comparing the VHDL output with the corresponding MATLAB calculated signal. Using this verification method, the VHDL design was verified post place and route (PAR) on several different Virtex family FPGAs.

<p>When doing recording work in the studio, disturbing sound must be removed. In this thesis, the purpose of this thesis is to formulate a mathematical equation, by using MATLAB to identify a system, then using the system to do cancellation of disturbing sound. The method of doing cancellation is to subtract the simulated output by the actual output, and then the disturbing sound was cancelled. The main thesis work will focus on the system identification, which is the process of determining the characteristic of an unknown system. Three systems were identified with the same model structure, which is linear (ARX) model. After finding out the model, the model quality must be evaluated. If the model is valid, there is a discussion if it is possible to run the mathematical equation in the real application, and how is the market today.</p>