The radio frequency power amplifier (RF-PA) within a digital wireless transmitter is a critical component regarding both the energy consumption and the signal quality. Especially due to today's broadband multicarrier modulation methods that generate signals with high peak-to-average power ratio, it is very hard to construct RF-PAs that achieve good energy efficiency and fulfill the strict linearity requirements imposed by the standard. Because of this, the digital predistortion (DPD) of RF-PAs has become a key technique for implementing energy efficient, high data rate wireless transmitters. This thesis investigates theoretical foundations and practical methods for the behavioral modeling and DPD of RF-PAs. The main contributions are a semi-physical model of the joint linearity-efficiency characteristics of RF-PAs, a detailed analysis of polynomial baseband models of RF-PAs focusing on the often neglected even-order terms in baseband, and a collection of practical methods for ...

Modern modulation methods as used in 3rd generation mobile communications (UMTS) generate strongly fluctuating transmission signal envelopes with high peak-to-average power ratios. These properties result in significant distortion due to the nonlinear behavior of the radio-frequency power amplifier (RF PA). We propose different nonlinear model structures for such amplifiers, based on memory polynomials and frequency-domain Volterra kernel expansion, where we can reduce the number of free parameters by 80% compared to traditional Volterra series approaches. Because these nonlinear models incorporate memory, we are able to model the nonlinear distortion of RF PAs with sufficient accuracy (e.g., −30 dB relative modeling error ), including the wideband case (bandwidth B = 20 MHz as needed for four-carrier WCDMA). Furthermore, we propose a method to construct RF PA models from frequency-dependent AM/AM and AM/PM conversions. For the compensation of the nonlinearities, we analyze ...

Audio effects modeling is the process of emulating an audio effect unit and seeks to recreate the sound, behaviour and main perceptual features of an analog reference device. Audio effect units are analog or digital signal processing systems that transform certain characteristics of the sound source. These transformations can be linear or nonlinear, time-invariant or time-varying and with short-term and long-term memory. Most typical audio effect transformations are based on dynamics, such as compression; tone such as distortion; frequency such as equalization; and time such as artificial reverberation or modulation based audio effects. The digital simulation of these audio processors is normally done by designing mathematical models of these systems. This is often difficult because it seeks to accurately model all components within the effect unit, which usually contains mechanical elements together with nonlinear and time-varying analog electronics. Most existing ...

This thesis focuses on different nonlinear distortion aspects in radio transmitter and receivers. Such nonlinear distortion aspects are generally becoming more and more important as the communication waveforms themselves get more complex and thus more sensitive to any distortion. Also balancing between the implementation costs, size, power consumption and radio performance, especially in multiradio devices, creates tendency towards using lower cost, and thus lower quality, radio electronics. Furthermore, increasing requirements on radio flexibility, especially on receiver side, reduces receiver radio frequency (RF) selectivity and thus increases the dynamic range and linearity requirements. Thus overall, proper understanding of nonlinear distortion in radio devices is essential, and also opens the door for clever use of digital signal processing (DSP) in mitigating and suppressing such distortion effects. On the receiver side, the emphasis in this thesis is mainly on the analysis and DSP ...

Modern signal processing applications emerging in telecommunication and instrumentation industries need high-speed analog-to-digital converters (ADCs), which can be achieved by employing a time-interleaved parallel array of ADCs (time-interleaved ADCs). The time interleaving of the channels allows to increase the sampling rate by the number of channels compared to a single channel. Unfortunately, time-interleaved ADCs suffer from channel mismatches that limit their performance, wherefore this thesis deals with the identification and compensation of channel mismatches in time-interleaved ADCs. By using nonlinear hybrid filter banks, we have modeled and analyzed channel mismatches in detail. The model covers linear and nonlinear channel mismatches, unifies, and extends the channel models found in the literature. A novel foreground channel mismatch identification method has been developed, which can be used to fully characterize dynamic linear mismatches. A background identification method provides accurate timing mismatch estimates. Finally, ...

The performance of today's communication systems is highly dependent on the employed analog-to-digital converters (ADCs), and in order to provide more flexibility and precision for the emerging communication technologies, high-performance ADCs are required. In this regard, the time-interleaved operation of an array of ADCs (TI-ADC) can be a reasonable solution. A TI-ADC can increase its throughput by using M channel ADCs or subconverters in parallel and sampling the input signal in a time-interleaved manner. However, the performance of a TI-ADC badly suffers from the mismatches among the channel ADCs. The mismatches among channel ADCs distort the TI-ADC output spectrum by introducing spurious tones besides the actual signal components. This thesis deals with the adaptive background calibration of frequency-response mismatches in a TI-ADC. By modeling each channel ADC as a linear time-invariant system, we develop the continuous-time, discrete-time, and time-varying system ...

To satisfy the continuously growing demands for higher data rates, modern radio communication systems employ larger bandwidths and more complex waveforms. Furthermore, radio devices are expected to support a rich mixture of standards such as cellular networks, wireless local-area networks, wireless personal area networks, positioning and navigation systems, etc. In general, a "smart'' device should be flexible to support all these requirements while being portable, cheap, and energy efficient. These seemingly conflicting expectations impose stringent radio frequency (RF) design challenges which, in turn, call for their proper understanding as well as developing cost-effective solutions to address them. The direct-conversion transceiver architecture is an appealing analog front-end for flexible and multi-standard radio systems. However, it is sensitive to various circuit impairments, and modern communication systems based on multi-carrier waveforms such as Orthogonal Frequency Division Multiplexing (OFDM) and Orthogonal Frequency Division Multiple ...

In the world of Information and communications technology (ICT), strategies for innovation and development are increasingly focusing on applications that require spatial representation and real-time interaction with and within 3D-media environments. One of the major challenges that such applications have to address is user-centricity, reflecting e.g. on developing complexity-hiding services so that people can personalize their own delivery of services. In these terms, multimodal interfaces represent a key factor for enabling an inclusive use of new technologies by everyone. In order to achieve this, multimodal realistic models that describe our environment are needed, and in particular models that accurately describe the acoustics of the environment and communication through the auditory modality are required. Examples of currently active research directions and application areas include 3DTV and future internet, 3D visual-sound scene coding, transmission and reconstruction and teleconferencing systems, to name but ...

In an automotive environment, quality of speech communication using a hands-free equipment is often deteriorated by interfering car noise. In order to preserve the speech signal without car noise, a multichannel speech enhancement system including a beamformer and a post-filter can be applied. Since employing a beamformer alone is insufficient to substantially reducing the level of car noise, a post-filter has to be applied to provide further noise reduction, especially at low frequencies. In this thesis, two novel post-filter designs along with their optimization for different driving conditions are presented. The first post-filter design utilizes an adaptive smoothing factor for the power spectral density estimation as well as a hybrid noise coherence function. The hybrid noise coherence function is a mixture of the diffuse and the measured noise coherence functions for a specific driving condition. The second post-filter design applies ...

With the advent of spectrally efficient wireless communication systems employing modulation schemes with varying amplitude of the communication signal, linearisation techniques for nonlinear microwave power amplifiers have gained significant interest. The availability of fast and cheap digital processing technology makes digital pre-distortion an attractive candidate as a means for power amplifier linearisation since it promises high power efficiency and fleexibility. Digital pre-distortion is further in line with the current efforts towards software defined radio systems, where a principal aim is to substitute costly and inflexible analogue circuitry with cheap and reprogrammable digital circuitry. Microwave power amplifiers are most efficient in terms of delivered microwave output power vs. supplied power if driven near the saturation point. In this operational mode, the amplifier behaves as a nonlinear device, which introduces undesired distortions in the information bear- ing microwave signal. These nonlinear distortions ...

Time-varying systems are encountered in various fields of engineering. If the time-varying behavior of a system is undesired, it produces a distorted output signal. Dedicated time-varying systems can be cascaded with the original system to correct the impact of the undesired time-varying behavior on the output signal. In applications where a high reconstruction accuracy is important, the computational cost of designing and employing flexible digital correction systems remains challenging. In particular, the computational load becomes a major challenge if the digital correction system needs to be redesigned for each time instant. In this thesis, low complexity correction methods for the design of linear time-varying correction filters are presented. These filters can be applied to postcorrect or precorrect linear time-varying systems. In order to mitigate the computational complexity of the filter design, a low complexity filter design algorithm for the least-squares ...

In the early days of speech synthesis research, understanding voice production has attracted the attention of scientists with the goal of producing intelligible speech. Later, the need to produce more natural voices led researchers to use prerecorded voice databases, containing speech units, reassembled by a concatenation algorithm. With the outgrowth of computer capacities, the length of units increased, going from diphones to non-uniform units, in the so-called unit selection framework, using a strategy referred to as 'take the best, modify the least'. Today the new challenge in voice synthesis is the production of expressive speech or singing. The mainstream solution to this problem is based on the “there is no data like more data” paradigm: emotionspecific databases are recorded and emotion-specific units are segmented. In this thesis, we propose to restart the expressive speech synthesis problem, from its original voice ...

The amount of digital music available both on the Internet and by each listener has considerably raised for about ten years. The organization and the accessibillity of this amount of data demand that additional informations are available, such as artist, album and song names, musical genre, tempo, mood or other symbolic or semantic attributes. Automatic music indexing has thus become a challenging research area. If some tasks are now correctly handled for certain types of music, such as automatic genre classification for stereotypical music, music instrument recoginition on solo performance and tempo extraction, others are more difficult to perform. For example, automatic transcription of polyphonic signals and instrument ensemble recognition are still limited to some particular cases. The goal of our study is not to obain a perfect transcription of the signals and an exact classification of all the instruments ...

In the recent years, a significant portion of the research within the field of wireless communications has been motivated by two aspects: the constant increase in the number of wireless devices and the higher and higher data rate requirements of the individual applications. The undisputed outcome of these phenomena is the heavy congestion of the suitable spectral resources. This has inspired many innovative solutions for improving the spectral efficiency of the wireless communications systems by facilitating more simultaneous connections and higher data rates without requiring additional spectrum. These include technologies such as in-phase/quadrature (I/Q) modulation, multiple-input and multiple-output (MIMO) systems, and the orthogonal frequency-division multiplexing (OFDM) waveform, among others. Even though these existing solutions have greatly improved the spectral efficiency of wireless communications, even more advanced techniques are needed for fulfilling the future data transfer requirements in the ultra high ...

In this thesis novel implementation approaches for standardized and non-standardized ultra wide-band (UWB) systems are presented. These implementation approaches include signal processing algorithms to achieve processing of UWB signals in transceiver front-ends and in digital back-ends. A parallelization of the transceiver in the frequency-domain has been achieved with hybrid filterbank transceivers. The standardized MB-OFDM signaling scheme allows par- allelization in the frequency domain by distributing the orthogonal multicarrier modulation onto multiple units. Furthermore, the channel’s response to wideband signals has been parallelized in the frequency domain and the effects of the parallelization have been investi- gated. Slight performance decreases are observed, where the limiting effects are truncated sidelobes and filter mismatches in analog front-ends. Measures for the performance loss have been defined. For UWB signal generation, a novel broadband signal generation approach is presented. For that purpose, multiple digital-to-analog converters ...