A presente tese descreve a construção de uma nova transformada wavelet, aqui chamada de SPIKELET, que, combinada com um algoritmo proposto, é aplicada no reconhecimento computacional de padrões em spikes (picos) e spikes sobrepostos (overlaps) encontrados no sinal digitalizado correspondente às reações do neurônio H1 do cérebro de uma mosca de ordem Diptera, que é sensível aos estímulos visuais do meio externo. O algoritmo fornece, além do formato do sinal encontrado, o \'\'instante\'\' em que ele ocorreu, sendo que a implementação é feita, inclusive, em tempo-real, com o uso de um DSP. / This thesis describes the construction of a new wavelet transform, that is called SPIKELET, which is used together with a proposed algorithm, for spikes and overlaps pattern recognition, in a digitalized signal corresponding to the H1 visual neuron action potential from a Diptera\'s fly brain. The algorithm provides both the shape of the identified signal and the \'\'instant\'\' of time it happened. The implementation is also done in real time, using a DSP.

A presente tese descreve a construção de uma nova transformada wavelet, aqui chamada de SPIKELET, que, combinada com um algoritmo proposto, é aplicada no reconhecimento computacional de padrões em spikes (picos) e spikes sobrepostos (overlaps) encontrados no sinal digitalizado correspondente às reações do neurônio H1 do cérebro de uma mosca de ordem Diptera, que é sensível aos estímulos visuais do meio externo. O algoritmo fornece, além do formato do sinal encontrado, o \'\'instante\'\' em que ele ocorreu, sendo que a implementação é feita, inclusive, em tempo-real, com o uso de um DSP. / This thesis describes the construction of a new wavelet transform, that is called SPIKELET, which is used together with a proposed algorithm, for spikes and overlaps pattern recognition, in a digitalized signal corresponding to the H1 visual neuron action potential from a Diptera\'s fly brain. The algorithm provides both the shape of the identified signal and the \'\'instant\'\' of time it happened. The implementation is also done in real time, using a DSP.

Yes. While the results presented in this thesis declare that listeners prefer the recording of an acoustic instrument, there is still a positive response about virtual violins, especially when compared to an unfiltered electric instrument. In this context a virtual violin is the result of convolving the raw signal from an electric instrument with the impulse from a real violin. A key part of this process is the characterisation of this impulse response, of which there are several methods. This thesis explores the use of virtual violins in acoustics research and music performance. The opening chapters provide an overview of the literature about violin acoustics and previous uses of virtual violins. A significant portion of the thesis details the development of a system that is used to produce digital characterisations of violins. The method used involved the measurement of sound radiation from the violin body after it had been excited by an impulse. This impulse is provided by an instrumented hammer, which strikes the violin on the bridge. One of the pitfalls of this method is the imperfect frequency response of the strike, which is corrected using a deconvolution algorithm. Deconvolution is an important part of the process and is discussed at length in the thesis. The described characterisation system utilised a bespoke frame that could hold the violin by the neck and rotate it to a specific angle on a single plane. This enabled the characterisation of a violin at incremental angles. Not only can these characterisations be used for the development of virtual violins, but they can also be used to analyse the spectral properties of the instrument. These characterisations at different angles allowed an examination of the violin's directivity, which explored how the sound energy is distributed from the instrument. Analysis determined that in the mid-range (C4-B5) the sound distribution is fairly isotropic. Outside of these ranges the sound distribution is significantly anisotropic. Finally, the thesis details three psychoacoustic experiments where participants listened to different audio samples and rated them according to their personal preference. The first of these had five virtual violins, of various manufacturers and ages, along with an unfiltered electric instrument. Listeners of mixed musical ability were asked to listen to recorded samples of these instruments and to provide preference scores for each. The experiment found that listeners preferred the sound of virtual violins to the unfiltered electric. The next experiment presented musically trained participants with samples of a virtual violin with varying impulse response lengths. This revealed that there was a sigmoid shaped relationship between the number of coefficients of the violin's impulse response and the listener's preference score. The final experiment was designed to examine the preference scores towards a recording of a real violin and its emulated equivalent. Participants of two groups, one with musical training and one without, took part in the study. The results showed that the real violin was preferred to the emulated violin and that musical training did not have any effect on the preference scores. The statistical analysis also determined that there was no interaction between violin treatment and musical training. An additional question was also posed to the participants post-experiment: Which of the presented samples is the real instrument? The results showed that those with musical training were significantly better at identifying the real instrument compared to those without such training. Speculation is drawn as to why this might be, though it is speculated that the spatial effects, such as those described by Weinreich, had no effect.

This master thesis presents the development and implementation of two digital systems based on Field-programmable Gate Array (FPGA): a Bit Error Rate Testing (BERT) system for an Optical Communication (OCOM) application, and a Digital Signal Processing (DSP) system for a Spin-Dependent Tunneling Magnetometer (SDTM). Both applications are intended for space and currently under development at the Ångström Space Technology Center (ÅSTC). The DSP system samples analog signals and applies a Fast Fourier Transform (FFT) for to provide frequency spectrum analysis. The report covers detailed system designs, state machine designs, and accounts for system verifications and measurements. As the live OCOM system and SDTM were unavailable by the time of testing, a series of emulated testing cases was set up to evaluate the digital systems developed. The BERT system was evaluated by checking the bit error rate of a stranded wire connection and a coaxial cable. Analog square and sine wave signals were used to evaluate the performance and accuracy of the FFT in the DSP system. Both systems were functionally verified using the Altera SignalTap II Logic Analyzer. Analysis of the measurement results for the testing cases indicates that the BERT works well at clock frequencies of 50 and 125 MHz, and that the coaxial cable is more suitable for data transmission as it gives a lower bit error rate than the stranded wire. The DSP system was verified to work well at a clock frequency of 62.5 MHz, and is able to sample any waveform at a sampling frequency of 62.5 MHz and continuously gets, at maximum, 14-bit wide digital signals. The sample point lengths for FFT are 64, 512 and 1024, and the data transfer rate between the FPGA and the computer reaches 115200 baud. In conclusion, the developed BERT and DSP system can be used to support the OCOM and the SDTM hardware, respectively.

Digital designs are often very large and complex, this makes locating and fixing a bug very hard and time consuming. Often more than half of the development time is spent on verification. Assertion based verification is a method that uses assertions that can help to improve the verification time. Simulating with assertions provides more information that can be used to locate and correct a bug. In this master thesis assertions are discussed and implemented in Senior DSP processor.

I dagens högteknologiska samhälle kryllar hemmen av datorer i olika former. Fler och fler använder en smartphone dagligen och inser kanske inte vilken datorkraft som finns inuti det de håller i handen. I detta arbete presenteras tillämpning av signalbehandlingsteknik (DSP) som har potentialen att på ett effektivt sätt sänka kostnaden för tillverkning av högtalare genom att kombinera dessa med ett filter som kontinuerligt kompenserar för högtalarens tillkortakommanden och skavanker i ljudåtergivning med hjälp av en dator.I detta arbete väljs komponenter och konstruktioner för högtalare inte bara efter prestanda. En avvägning görs mellan å ena sidan vad som gör högtalaren så billig som möjligt att bygga, och å andra sidan vilka av konstruktionens prestanda som är möjliga att optimera i efterhand med DSP. Därefter används digital signalbehandling för att kompensera de brister som finns och på så sätt återskapa den ljudkvalitet som förväntas av högtalaren.I denna rapport beskrivs arbetet med en prototyp som är framtagen enkom för användning tillsammans med DSP. Prototypen färdigställs tillräckligt för att den ska kunna analyseras och utvärderas. En redan existerande produkt från uppdragsgivaren XTZ AB får ett nytt anpassat delningsfilter för DSP. Mätningar och lyssningstester avslutar sedan rapporten och en bedömning görs huruvida tekniken fungerar tillfredsställande för ändamålet och har potential att på ett brett sätt användas inom hifi-sammanhang.De filter som används kräver inte mer beräkningskraft av en dator än att de skulle kunna användas i en mobiltelefon som sedan kan dockas i en stereo och därmed får ut det mesta av anläggningen, eller att en vanlig bärbar dator kopplas till ljudanläggningen och med ett knapptryck kompenserar för de ojämnheter som finns i ljudåtergivningen. / This thesis presents an application of digital signal processing (DSP) techniques that have the potential to lower the production cost for loudspeakers. Combining a speaker construction with matched digital filters that continuously compensate the signal to remedy the shortcomings of the construction allows cost cuts in production of components and selection of drivers. A started prototype is finished to the extent that it can be measured and used evaluate the potential of the applied technology. Further, an existing product is reconstructed in order to to cut production costs and is then compared to its original. After measurements and listening tests have been performed, it is concluded that the technology has great potential to become a key component in cutting speaker design and production costs.

This paper presented Symmetric and Asymmetric Regular Sampling techniques apply successfully to a Digital Signal Processor (DSP) together with an inverter. Experimental results are shown that taking advantage of SPWM control techniques can reduce the waste of energy. Further, the Soft Start techniques of DSP scripts can boost security for driver. Moreover motor driver safely can be increased by designing circuit to prevent over voltage. The function of motor driver will be better due to the decreased hardware size and increased accuracy which are the advantage of writing DSP scripts to analysis rotor speed.

In this thesis, we present an implementation of a low-power digital signal processor. We design the hardware units and analyze the instruction set for digital signal process applications. Besides, the power consumption issue is considered. We present two solutions to reduce the power consumption. We also discuss the test pattern generations to verify this DSP processor. Finally, the concept of IP design is considered in this design.

The thesis is accomplished two DSP-based On-Line UPS using voltage and current control to implement the parallel operation. Each inverter in the parallel operation system has the same control method. The system can reduce the zero crossover distribution causing by SPWM and the influence causing by load variation with inner current and outer voltage loop control which are taking advantage of P and PI control respectively. On the premise of the same system parameters, the two system can achieve the synchronous between phase and frequency by the control of digital PLL circuit to eliminate the circulation current.

The thesis is one DSP-based On-Line UPS with USB2.0 communication interface and monitor software. The system is consisting of three sub-systems. The sub-system one is the DSP development system:First step, it uses the DSP to generate the PWM signal to inverter.Second step,ultilizes the ADC module of the DSP to implement the P type current controller in the inner feedback loop and P-I type voltage controller in the outer feedback loop.Third step,accomplishes the synchronous detection via ETU. And then transfer the UPS system parameters to USB2.0 development board through EMI.The sub-system two is the USB2.0 development system and the monitor software:The function of this sub-system is to transmit the parameters from DSP to PC in the specification of USB2.0.In order to achieve the goal, the firmware is required for the USB development board.Once the monitor software is activated, the firmware is downloaded to the board.The sub-system three is the electronic devices:like inverter¡Bconverter¡Bpfc¡Bpower board¡Bbattery etc.Those devices constitute the fundamental hardware circuits modules of the UPS.