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>

Fiber-optic networks are continually evolving to accommodate ever-increasing data transport rates demanded by modern applications, devices, and services. Network operators are now beginning to deploy systems with 100 Gb/s per-wavelength data rates while maintaining the 50 GHz dense wavelength division multiplexing grid that is (generally) standard for 10 Gb/s systems. Advanced modulation formats incorporating both amplitude- and phase-based data symbols are necessary to meet the spectral efficiency requirements of fiber-optic data transport. These modulation formats require coherent detection, enabling future networks to take advantage of advances in silicon CMOS via digital signal processing algorithms and techniques. The primary challenge for future networks is the fiber nonlinear response; changes in the intensity of the propagating optical signal induce changes in the optical fiber refractive index. Limiting the allowed propagation intensity will reduce these nonlinear effects and correspondingly limit the total available signal-to-noise ratio (SNR) within the channel. Predicting the nonlinear SNR limits of fiber-optic transport for data rates 100 Gb/s and beyond is a primary purpose of this research. This dissertation expressly matches several novel expressions for nonlinear interference accumulation to experimental data and demonstrates robust theoretical prediction of nonlinear transmission penalties. The experiments were performed to isolate the transmission performance of the fiber medium in the highly dispersive regime -- no dispersion compensation or Raman amplification was employed and all other hardware was kept static. These results are the first experimental validation of the nonlinear interference expressions on a fiber-type basis. Second, this dissertation moves to data transport beyond per-wavelength rates of 100 Gb/s by employing 16QAM at baud rates as high as 32 GHz. It examines signal processing strategies for 16QAM transport and extends the nonlinear interference prediction techniques to 16QAM. The results reveal that the SNR requirements of 16QAM as limited by nonlinear interference will likely limit deployments to high-density regional and metro networks.

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.

State of the art model based coders offer a perceptually acceptable reconstructed speech quality at bit-rates as low as 2000 bits per second. However, the performance of these coders rapidly deteriorates below this rate, primarily since very few bits are available to encode the model parameters with high fidelity. This thesis aims to meet the challenge of designing speech coders that operate at lower bit-rates while reconstructing the speech at the receiver at the same or even better quality than state of the art low bit-rate speech coders. In one of the contributions, we develop a plethora of techniques for efficient coding of the parameters obtained by the MELP algorithm, under the assumption that the classification of the frames of the MELP coder is available. Also, a simple and elegant procedure called dynamic codebook reordering is presented for use in the encoders and decoders of a vector quantization system that effectively exploits the correlation between vectors of parameters obtained from consecutiv speech frames without introducing any delay, distortion or suboptimality. The potential of this technique in significantly reducing the bit-rates of speech coders is illustrated. Additionally, the thesis also attempts to address the issues of designing such very low bit-rate speech coders so that they are robust to environmental noise. To impart robustness, a speech enhancement framework employing Kalman filters is presented. Kalman filters designed for speech enhancement in the presence of noise assume an autoregressive model for the speech signal. We improve the performance of Kalman filters in speech enhancement by constraining the parameters of the autoregressive model to belong to a codebook trained on clean speech. We then extend this formulation to the design of a novel framework, called the multiple input Kalman filter, that optimally combines the outputs from several speech enhancement systems. Since the low bit-rate speech coders compress the parameters significantly, it is very important to protect the transmitted information from errors in the communication channel. In this thesis, a novel channel-optimized multi-stage vector quantization codec is presented, in which the stage codebooks are jointly designed.

This paper proposes a sensorless control construction adapting to different speed with DSP2407 as the signal processing control core for rotation compressor. The sensorless control method obtains the rotor position by detecting the back electromotive force signals directly, then obtains better communications and the speed estimation by using digital signal which controlling power switches. Finally ,it carries out speed feedback control and current feedback control to improvt efficiency. Comparing adaptive-step control with traditional-step control and six-step control , the experiment result shows that adaptive-step control has better efficiency and lower vibration.

Electric power utilities and end users are becoming increasingly concerned about the quality of power supply. To reduce the losses caused by power service disturbances, mitigation devices are available for improving the power quality. The first step in the power quality improvement is to monitor the system behavior by using some Electronic recording devices. A design and implementation of a digital signal processor based power quality monitoring device is presented in this thesis. Several event-triggering methods are studied and implemented to detect system disturbances. Simulation and test results indicate that the proposed design can meet the requirements for power measurements and transient event recording during steady and transient states.

The DSP is used as the control kernel in this thesis, proposing a method of sensorless and variable speed driving with current feedback for the DC compressor. By detecting the back electromotive force signals directly, the information of rotor position can be obtained, the commutation process and the speed estimation can also be achieved. Combining the current feedback method, the sinusoidal commutation with sensorless control makes the motor lower speed ripple and higher rotating efficiency. The results show that the sinusoidal commutation approach has the advantages of higher efficiency and less speed ripple as compared to the approaches of traditional-step commutation and six-step with current feedback by experimental setting.