Syftet med arbetet har varit att nå utvecklad kännedom inom vokalproduktion genom att undersöka vad som kan göras för att höja kvaliteten på sången i en musikproduktion. En viktig fråga har varit om fokus, för att nå ett gott resultat, framförallt vikten på inspelningen av sången eller om mycket också kan göras genom efterbearbetning för att nå ett gott resultat. Under detta projekt har jag valt att arbeta med tre produktioner där min egen sång står i centrum. Dessa har lite olika stilar på sången och jag har valt just dessa för att utmana mig själv och se hur de olika sångerna ska poleras för att få sin plats i en slutgiltig produktion. För produktionen har jag använt mig av Logic Pro X som DAW. Resultatet är tre produktioner där jag bedömer att sången genom de produktionsmetoder som tillämpats har blivit betydligt bättre jämfört med tidigare produktioner som jag har genomfört. Bland de metoder och tekniker som jag tillämpat ingår programmet Melodyne som jag använt för att höja sångens kvalitet genom att sången blivit tightare och renare. Även equalizer ochkompressor har bidragit till att få sången att låta jämn. En viktig slutsats är att all sång, trots omfattande bearbetning med effektiva redigeringsverktyg, skulle behöva sjungas om innan de aktuella låtarna kan publiceras genom ett eventuellt släpp. / The aim of the work has been to gain advanced knowledge in vocal production by investigating different methods to raise the quality of the vocal parts in a music production. An important question has been whether the focus, to achieve a good result, mainly has to be the recording of the song or whether much can be done through post-processing in order to achieve a good result. During this project I have chosen to work with three productions where I sing the vocal parts myself. These have slightly different styles on the song and I have chosen just these to challenge myself and how to edit and produce the vocal part for at good result in the final production. For the production I have used Logic Pro X as DAW. The result is three productions where the production methods used, from my view, clearly has contributed to a better result compared with productions that I previously have performed. The methods and techniques that I have applied include the Melodyne software, which I have used to improve the quality of the song by making the song tighter and cleaner. Also equalizer and compressor have been used to improve the sound quality. However, it is important to conclude that the vocal parts in the productions, despite extensive processing with efficient editing tools, would need to be re-recorded before the songs can be published.

This paper outlines the design of a high-precision graphic audio equalizer with digital filters in parallel, along with its implementation in MATLAB App Designer. The equalizer is comprised of 31 bands separated with a one-third octave frequency ratio, and its frequency response is controlled by 63 filters. Furthermore, the application can process audio signals, in real time, recorded by microphone and from audio files. While processing, it displays an FFT plot of the output sound, also in real time, equipped with a knob by which the refreshing pace can be adjusted. The actual frequency response proved to match the desired one accurately, but the matching is computationally demanding for the computer. An even higher accuracy would entail a computational complexity beyond the power of ordinary computers, and was thus concluded to be inappropriate. As a result, the final application manages to provide most laptops with both high precision and proper functionality.

Tato práce je psaná v angličtině a je zaměřená na problematiku ekvalizace v digitálních komunikačních systémech. Teoretická část zahrnuje stručné pozorování různých způsobů návrhu ekvalizérů. Praktická část se zabývá implementací nejčastěji používaných ekvalizérů a s jejich adaptačními algoritmy. Cílem praktické části je porovnat jejich charakteristiky a odhalit činitele, které ovlivňují kvalitu ekvalizace. V rámci problematiky ekvalizace jsou prozkoumány tři typy ekvalizérů. Lineární ekvalizér, ekvalizér se zpětnou vazbou a ML (Maximum likelihood) ekvalizér. Každý ekvalizér byl testován na modelu, který simuloval reálnou přenosovou soustavu s komplexním zkreslením, která je složena z útlumu, mezisymbolové interference a aditivního šumu. Na základě implenentace byli určeny charakteristiky ekvalizérů a stanoveno že optimální výkon má ML ekvalizér. Adaptační algoritmy hrají významnou roli ve výkonnosti všech zmíněných ekvalizérů. V práci je nastudována skupina stochastických algoritmů jako algoritmus nejmenších čtverců(LMS), Normalizovaný LMS, Variable step-size LMS a algoritmus RLS jako zástupce deterministického přístupu. Bylo zjištěno, že RLS konverguje mnohem rychleji, než algoritmy založené na LMS. Byly nastudovány činitele, které ovlivnili výkon popisovaných algoritmů. Jedním z důležitých činitelů, který ovlivňuje rychlost konvergence a stabilitu algoritmů LMS je parametr velikosti kroku. Dalším velmi důležitým faktorem je výběr trénovací sekvence. Bylo zjištěno, že velkou nevýhodou algoritmů založených na LMS v porovnání s RLS algoritmy je, že kvalita ekvalizace je velmi závislá na spektrální výkonové hustotě a a trénovací sekvenci.

Orthogonal frequency division multiplexing (OFDM) modulator with redundancy has been adopted in many wireless communication systems for higher data rate transmissions. The introduced redundancy at the transmitter allows us to overcome serious inter-block interference (IBI) problems due to highly dispersive channel. However, the selection of redundancy length will affect the system performance and spectral efficiency, and is highly dependent on the length of channel impulse response. In this thesis, based on the pseudorandom postfix (PRP) OFDM scheme we propose a novel block-based OFDM transceiver framework. Since in the PRP-OFDM system the PRP can be employed for semi-blind channel estimation with order-one statistics of the received signal. Hence, for sufficient redundancy case the PRP-OFDM system with the Bayesian decision feedback equalizer (DFE) is adopted for suppressing the IBI and ISI simultaneously. However, for the insufficient redundancy case (the length of redundancy is less than the order of channel), we first propose a modified scheme for channel estimation. To further reduce the complexity of receiver, the maximum shortening signal-to-noise-ratio time domain equalizer (MSSNR TEQ) with the Bayesian DFE is developed for suppressing the IBI and ISI, separately. That is, after knowing the channel state information (CSI) and removing the effect of IBI with MSSNR TEQ, the Bayesian DFE is applied for eliminating the ISI. Via computer simulation, we verify that performance improvement, in terms of bit error rate (BER), compared with the conventional block-based minimum mean square error (MMSE)-DFE can be achieved.

This thesis proposes a decision-feedback equalizer (DFE) scheme for blind ADC-based receivers to overcome the challenges introduced by blind sampling. It presents the design, simulation, and implementation of a 5Gb/s speculative DFE for a 2x blind ADC-based receiver. The complete receiver, including the ADC, the DFE, and a 2x blind clock and data recovery (CDR) circuit, is implemented in Fujitsu’s 65-nm CMOS process. Measurements of the fabricated test-chip confirm 5Gb/s data recovery with bit error rate (BER) less than 1e−12 in the presence of a test channel introducing 13.3dB of attenuation at the Nyquist frequency of 2.5GHz. The receiver tolerates 0.24UIpp of high-frequency sinusoidal jitter (SJ) in this case. Without the DFE, the BER exceeds 1e−8 even when no SJ is applied.

This thesis proposes a decision-feedback equalizer (DFE) scheme for blind ADC-based receivers to overcome the challenges introduced by blind sampling. It presents the design, simulation, and implementation of a 5Gb/s speculative DFE for a 2x blind ADC-based receiver. The complete receiver, including the ADC, the DFE, and a 2x blind clock and data recovery (CDR) circuit, is implemented in Fujitsu’s 65-nm CMOS process. Measurements of the fabricated test-chip confirm 5Gb/s data recovery with bit error rate (BER) less than 1e−12 in the presence of a test channel introducing 13.3dB of attenuation at the Nyquist frequency of 2.5GHz. The receiver tolerates 0.24UIpp of high-frequency sinusoidal jitter (SJ) in this case. Without the DFE, the BER exceeds 1e−8 even when no SJ is applied.

ITC/USA 2006 Conference Proceedings / The Forty-Second Annual International Telemetering Conference and Technical Exhibition / October 23-26, 2006 / Town and Country Resort & Convention Center, San Diego, California / Many satellite payloads require wide-band channels for transmission of large amounts of data to users on the ground. These channels typically have substantial distortions, including bandlimiting distortions and high power amplifier (HPA) nonlinearities that cause substantial degradation of bit error rate performance compared to additive white Gaussian noise (AWGN) scenarios. An adaptive equalization algorithm has been selected as the solution to improving bit error rate performance in the presence of these channel distortions. This paper describes the design and implementation of an adaptive baseband equalizer (ABBE) utilizing the latest FPGA technology. Implementation of the design was arrived at by first constructing a high fidelity channel simulation model, which incorporates worst-case signal impairments over the entire data link. All of the modem digital signal processing functions, including multirate carrier and symbol synchronization, are modeled, in addition to the adaptive complex baseband equalizer. Different feedback and feed-forward tap combinations are considered as part of the design optimization.

International Telemetering Conference Proceedings / October 18-21, 2004 / Town & Country Resort, San Diego, California / This paper will present the results of experiments to characterize the performance of a blind, adaptive constant modulus algorithm (CMA) equalizer in simulated telemetry multipath environments. The variables included modulation method, bit rate, received signal-to-noise ratio, delay of the indirect path relative to the direct path, amplitude of the indirect path relative to the direct path, and fade rate. The main measured parameter was bit error probability (BEP). The tests showed that the equalizer usually improved the data quality in the presence of multipath.

International Telemetering Conference Proceedings / October 23-26, 2000 / Town & Country Hotel and Conference Center, San Diego, California / Design and performance evaluation of a low-complexity equalizer for recently standardized spectral efficient Feher patented quadrature phase shift keying (FQPSK) system [1] over multipath fading channel is presented. The implementation based on a Feher patented equalizer (FE) [1] is of a structure with three branches, which are individually used to compensate for a moving fade notch with different locations. These branches are switched by the control signal that is generated based on pseudo-error on-line detection technique. It is demonstrated that for typical aeronautical telemetry RF frequency selective fading channels, having delay spreads in 20 – 200 ns range, the adaptive FE reduces the number of statistical outages by more than 60% without the need for training bits and without increasing the receiver synchronization time.

The overwhelming trend of the usage of multimedia services has raised the consumers' awareness about quality. Both service providers and consumers are interested in the delivered level of perceptual quality. The perceptual quality of an original video signal can get degraded due to compression and due to its transmission over a lossy network. Video quality assessment (VQA) has to be performed in order to gauge the level of video quality. Generally, it can be performed by following subjective methods, where a panel of humans judges the quality of video, or by using objective methods, where a computational model yields an estimate of the quality. Objective methods and specifically No-Reference (NR) or Reduced-Reference (RR) methods are preferable because they are practical for implementation in real-time scenarios. This doctoral thesis begins with a review of existing approaches proposed in the area of NR image and video quality assessment. In the review, recently proposed methods of visual quality assessment are classified into three categories. This is followed by the chapters related to the description of studies on the development of NR and RR methods as well as on conducting subjective experiments of VQA. In the case of NR methods, the required features are extracted from the coded bitstream of a video, and in the case of RR methods additional pixel-based information is used. Specifically, NR methods are developed with the help of suitable techniques of regression using artificial neural networks and least-squares support vector machines. Subsequently, in a later study, linear regression techniques are used to elaborate the interpretability of NR and RR models with respect to the selection of perceptually significant features. The presented studies on subjective experiments are performed using laboratory based and crowdsourcing platforms. In the laboratory based experiments, the focus has been on using standardized methods in order to generate datasets that can be used to validate objective methods of VQA. The subjective experiments performed through crowdsourcing relate to the investigation of non-standard methods in order to determine perceptual preference of various adaptation scenarios in the context of adaptive streaming of high-definition videos. Lastly, the use of adaptive gain equalizer in the modulation frequency domain for speech enhancement has been examined. To this end, two methods of demodulating speech signals namely spectral center of gravity carrier estimation and convex optimization have been studied.