Design and implementation of an ETSI-SDR OFDM transmitter with power amplifier linearizer

Satellite radio has attained great popularity because of its wide range of geographical coverage and high signal quality as compared to the terrestrial broadcasts. Most Satellite Digital Radio (SDR) based systems favor multi-carrier transmission schemes, especially, orthogonal frequency division multiplexing (OFDM) transmission because of high data transfer rate and spectral efficiency. It is a challenging task to find a suitable platform that supports fast data rates and superior processing capabilities required for the development and deployment of the new SDR standards. Field programmable gate array (FPGA) devices have the potential to become suitable development platform for such standards. Another challenging factor in SDR systems is the distortion of variable envelope signals used in OFDM transmission by the nonlinear RF power amplifiers (PA) used in the base station transmitters. An attractive option is to use a linearizer that would compensate for the nonlinear effects of the PA. In this research, an OFDM transmitter, according to European Telecommunications Standard Institute (ETSI) SDR Technical Specifications 2007-2008, was designed and implemented on a low-cost Xilinx FPGA platform. A weakly nonlinear PA, operating in the L-band SDR frequency (1.450-1.490GHz), was used for signal transmission. An FPGA-based, low-cost, adaptive linearizer was designed and implemented based on the digital predistortion (DPD) reference design from Xilinx, to correct the distortion effects of the PA on the transmitted signal

Hardward and algorithm architectures for real-time additive synthesis

Additive synthesis is a fundamental computer music synthesis paradigm tracing its origins to the work of Fourier and Helmholtz. Rudimentary implementation linearly combines harmonic sinusoids (or partials) to generate tones whose perceived timbral characteristics are a strong function of the partial amplitude spectrum. Having evolved over time, additive synthesis describes a collection of algorithms each characterised by the time-varying linear combination of basis components to generate temporal evolution of timbre. Basis components include exactly harmonic partials, inharmonic partials with time-varying frequency or non-sinusoidal waveforms each with distinct spectral characteristics. Additive synthesis of polyphonic musical instrument tones requires a large number of independently controlled partials incurring a large computational overhead whose investigation and reduction is a key motivator for this work. The thesis begins with a review of prevalent synthesis techniques setting additive synthesis in context and introducing the spectrum modelling paradigm which provides baseline spectral data to the additive synthesis process obtained from the analysis of natural sounds. We proceed to investigate recursive and phase accumulating digital sinusoidal oscillator algorithms, defining specific metrics to quantify relative performance. The concepts of phase accumulation, table lookup phase-amplitude mapping and interpolated fractional addressing are introduced and developed and shown to underpin an additive synthesis subclass - wavetable lookup synthesis (WLS). WLS performance is simulated against specific metrics and parameter conditions peculiar to computer music requirements. We conclude by presenting processing architectures which accelerate computational throughput of specific WLS operations and the sinusoidal additive synthesis model. In particular, we introduce and investigate the concept of phase domain processing and present several “pipeline friendly” arithmetic architectures using this technique which implement the additive synthesis of sinusoidal partials

2020 Sound

Finlay Braithwaite, OCAD University, 2020 Sound, Master of Design (MDes), Digital Futures, 2019 2020 Sound is a positional-tracking microphone and DSP time-of-arrival alignment system. In production, an ultrasonic beacon emits a temporal positional reference that is captured by standard audio recording devices. This reference gives purchase to align multiple microphone perspectives of a source, correcting for their initial offset as well their movement throughout the recording. In capturing a sound source with multiple microphones, misaligned and drifting time-of-arrival of the source at each microphone greatly impacts the cohesion, focus, and impact of their summation in the mixing process. The common boom and lavaliere microphone scenario implemented in film and television production suffers from this misalignment and, as a result, time-intensive and inaccurate manual editorial processes are employed to align microphones before their summation. This system could remedy a fundamental issue encountered in audio production with the ultimate aim of improving the clarity and quality of productions that make use of 2020 Sound

Modeling and frequency tracking of marine mammal whistle calls

Submitted in partial fulfillment of the requirements for the degree of Master of Science at the Massachusetts Institute of Technology and the Woods Hole Oceanographic Institution February 2009Marine mammal whistle calls present an attractive medium for covert underwater communications. High quality models of the whistle calls are needed in order to synthesize natural-sounding whistles with embedded information. Since the whistle calls are composed of frequency modulated harmonic tones, they are best modeled as a weighted superposition of harmonically related sinusoids. Previous research with bottlenose dolphin whistle calls has produced synthetic whistles that sound too “clean” for use in a covert communications system. Due to the sensitivity of the human auditory system, watermarking schemes that slightly modify the fundamental frequency contour have good potential for producing natural-sounding whistles embedded with retrievable watermarks. Structured total least squares is used with linear prediction analysis to track the time-varying fundamental frequency and harmonic amplitude contours throughout a whistle call. Simulation and experimental results demonstrate the capability to accurately model bottlenose dolphin whistle calls and retrieve embedded information from watermarked synthetic whistle calls. Different fundamental frequency watermarking schemes are proposed based on their ability to produce natural sounding synthetic whistles and yield suitable watermark detection and retrieval

Investigation of Different Video Compression Schemes Using Neural Networks

Image/Video compression has great significance in the communication of motion pictures and still images. The need for compression has resulted in the development of various techniques including transform coding, vector quantization and neural networks. this thesis neural network based methods are investigated to achieve good compression ratios while maintaining the image quality. Parts of this investigation include motion detection, and weight retraining. An adaptive technique is employed to improve the video frame quality for a given compression ratio by frequently updating the weights obtained from training. More specifically, weight retraining is performed only when the error exceeds a given threshold value. Image quality is measured objectively, using the peak signal-to-noise ratio versus performance measure. Results show the improved performance of the proposed architecture compared to existing approaches. The proposed method is implemented in MATLAB and the results obtained such as compression ratio versus signalto- noise ratio are presented

Speech analysis and synthesis using an auditory model

Many traditional speech analysis/synthesis techniques are designed to produce speech with a spectrum that is as close as possible to the original. This may not be necessary because the auditory nerve is the only link from the auditory periphery to the brain, and all information that is processed by the higher auditory system must exist in the auditory nerve firing patterns. Rather than matching the synthesised speech spectra to the original representation, it should be sufficient that the representations of the synthetic and original speech be similar at the auditory nerve level. This thesis develops a speech analysis system that incorporates a computationally efficient model of the auditory periphery. Timing-synchrony information is employed to exploit the in-synchrony phenomena observed in neuron firing patterns to form a nonlinear relative spectrum intensity measure. This measure is used to select specific dominant frequencies to reproduce the speech based on a synthesis-by-sinusoid approach. The resulting speech is found to be intelligible even when only a fraction of the original frequencies are selected for synthesis. Additionally, the synthesised speech is highly noise immune, and exhibits noise reduction due to the coherence property of the frequency transform algorithm, and the dominance effect of the spectrum intensity measure. This noise reduction and low bit rate potential of the speech analysis system is exploited to produce a highly noise immune synthesis that outperforms similar representations formed both by a more physiologically accurate model and a classical non-biological speech processing algorithm. Such a representation has potential application in low-bit rate systems, particularly as a front end to an automatic speech recogniser

An investigation into the requirements for an efficient image transmission system over an ATM network

This thesis looks into the problems arising in an image transmission system when transmitting over an A TM network. Two main areas were investigated: (i) an alternative coding technique to reduce the bit rate required; and (ii) concealment of errors due to cell loss, with emphasis on processing in the transform domain of DCT-based images. [Continues.

Transmitter architectures with digital modulators, D/A converters and switching-mode power amplifiers

This thesis is composed of nine publications and an overview of the research topic, which also summarises the work. The research described in this thesis focuses on research into the digitalisation of wireless communication base station transmitters. In particular it has three foci: digital modulation, D/A conversion and switching-mode power amplification. The main interest in the implementation of these circuits is in CMOS. The work summarizes the designs of several circuit blocks of a wireless transmitter base station. In the baseband stage, a multicarrier digital modulator that combines multiple modulated signals at different carrier frequencies digitally at baseband, and a multimode digital modulator that can be operated for three different communications standards, are implemented as integrated circuits. The digital modulators include digital power ramping and power level control units for transmission bursts. The upconversion of the baseband signal is implemented using an integrated digital quadrature modulator. The work presented provides insight into the digital-to-analogue interface in the transmitters. This interface is studied both by implementing an intermediate frequency D/A converter in BiCMOS technology and bandpass Delta-Sigma modulator-based D/A conversion in CMOS technology. Finally, the last part of the work discusses switching-mode power amplifiers which are experimented with both as discrete and integrated implementations in conjunction with 1-bit Delta-Sigma modulation and pulse-width modulation as input signal generation methods.Tämä väitöskirja koostuu yhdeksästä julkaisusta ja tutkimusaiheen yhteenvedosta. Väitöskirjassa esitetty tutkimus keskittyy langattaman viestinnän tukiasemien lähettimien digitalisoinnin tutkimukseen. Yksityiskohtaisemmin tutkimusalueet ovat: digitaalinen modulaatio, D/A muunnos ja kytkinmuotoiset tehovahvistimet. Näiden elektronisten piirien toteutuksessa keskitytään CMOS teknologiaan. Työ vetää yhteen useiden langattoman viestinnän tukiasemien lähettimien piirilohkojen suunnittelun. Kantataajuusasteella toteutetaan integroituna piirinä monikantoaaltoinen digitaalinen modulaattori, joka yhdistää useita moduloituja signaaleja eri kantoaalloilla digitaalisesti ja monistandardi digitaalinen modulaatori, joka tukee kolmea eri viestintästandardia. Digitaaliset modulaattoripiirit sisältävät digitaalisen tehoramping ja tehotason säätöyksikön lähetyspurskeita varten. Kantataajuussignaalin ylössekoitus toteutetaan integroitua digitaalista kvadratuurimodulaattoria käyttäen. Esitetty työ antaa näkemystä lähettimien digitalia-analogia rajapintaan, jota tutkitaan toteuttamalla välitaajuinen D/A muunnin BiCMOS teknologialla ja päästökaistainen Delta-Sigma-modulaattoripohjainen D/A muunnin CMOS teknologialla. Lopuksi työn viimeinen osa käsittelee kytkinmuotoisia tehovahvistimia, joita tutkitaan kokeellisesti sekä erilliskompontein toteutettuina piirein että integroiduin piirein toteutettuina käyttäen sisääntulosignaalin muodostamismenetemänä yksibittistä Delta-Sigma-modulaatiota ja pulssin leveys modulaatiota.reviewe

頭部伝達関数の空間領域特性モデリング

Tohoku University鈴木陽一課