Wavelet Filter Banks Using Allpass Filters

Allpass filter is a computationally efficient versatile signal processing building block. The interconnection of allpass filters has found numerous applications in digital filtering and wavelets. In this chapter, we discuss several classes of wavelet filter banks by using allpass filters. Firstly, we describe two classes of orthogonal wavelet filter banks composed of two real allpass filters or a complex allpass filter, and then consider design of orthogonal filter banks without or with symmetry, respectively. Next, we present two classes of filter banks by using allpass filters in lifting scheme. One class is causal stable biorthogonal wavelet filter bank and another class is orthogonal wavelet filter bank, all with approximately linear phase response. We also give several design examples to demonstrate the effectiveness of the proposed method

Database of audio records

Diplomka a prakticky castDiplome with partical part

Realization of Integrable Low- Voltage Companding Filters for Portable System Applications

Undoubtedly, today’s integrated electronic systems owe their remarkable performance primarily to the rapid advancements of digital technology since 1970s. The various important advantages of digital circuits are: its abstraction from the physical details of the actual circuit implementation, its comparative insensitiveness to variations in the manufacturing process, and the operating conditions besides allowing functional complexity that would not be possible using analog technology. As a result, digital circuits usually offer a more robust behaviour than their analog counterparts, though often with area, power and speed drawbacks. Due to these and other benefits, analog functionality has increasingly been replaced by digital implementations. In spite of the advantages discussed above, analog components are far from obsolete and continue to be key components of modern electronic systems. There is a definite trend toward persistent and ubiquitous use of analog electronic circuits in day-to-day life. Portable electronic gadgets, wireless communications and the widespread application of RF tags are just a few examples of contemporary developments. While all of these electronic systems are based on digital circuitry, they heavily rely on analog components as interfaces to the real world. In fact, many modern designs combine powerful digital systems and complementary analog components on a single chip for cost and reliability reasons. Unfortunately, the design of such systems-on-chip (SOC) suffers from the vastly different design styles of analog and digital components. While mature synthesis tools are readily available for digital designs, there is hardly any such support for analog designers apart from wellestablished PSPICE-like circuit simulators. Consequently, though the analog part usually occupies only a small fraction of the entire die area of an SOC, but its design often constitutes a major bottleneck within the entire development process. Integrated continuous-time active filters are the class of continuous-time or analog circuits which are used in various applications like channel selection in radios, anti-aliasing before sampling, and hearing aids etc. One of the figures of merit of a filter is the dynamic range; this is the ratio of the largest to the smallest signal that can be applied at the input of the filter while maintaining certain specified performance. The dynamic range required in the filter varies with the application and is decided by the variation in strength of the desired signal as well as that of unwanted signals that are to be rejected by the filter. It is well known that the power dissipation and the capacitor area of an integrated active filter increases in proportion to its dynamic range. This situation is incompatible with the needs of integrated systems, especially battery operated ones. In addition to this fundamental dependence of power dissipation on dynamic range, the design of integrated active filters is further complicated by the reduction of supply voltage of integrated circuits imposed by the scaling down of technologies to attain twin objective of higher speed and lower power consumption in digital circuits. The reduction in power consumption with decreasing supply voltage does not apply to analog circuits. In fact, considerable innovation is required with a reduced supply voltage even to avoid increasing power consumption for a given signal to noise ratio (S/N). These aspects pose a great hurdle to the active filter designer. A technique which has attracted the attention of circuit designers as a possible route to filters with higher dynamic range per unit power consumption is “companding”. Companding (compression-expansion) filters are a very promising subclass of continuous-time analog filters, where the input (linear) signal is initially compressed before it will be handled by the core (non-linear) system. In order to preserve the linear operation of the whole system, the non-linear signal produced by the core system is converted back to a linear output signal by employing an appropriate output stage. The required compression and expansion operations are performed by employing bipolar transistors in active region or MOS transistors in weak inversion; the systems thus derived are known as logarithmic-domain (logdomain) systems. In case MOS transistors operated in saturation region are employed, the derived structures are known as Square-root domain systems. Finally, the third class of companding filters can also be obtained by employing bipolar transistors in active region or MOS transistors in weak inversion; the derived systems are known as Sinh-domain systems. During the last several years, a significant research effort has been already carried out in the area of companding circuits. This is due to the fact that their main advantages are the capability for operation in low-voltage environment and large dynamic range originated from their companding nature, electronic tunability of the frequency characteristics, absence of resistors and the potential for operations in varied frequency regions.Thus, it is obvious that companding filters can be employed for implementing high-performance analog signal processing in diverse frequency ranges. For example, companding filters could be used for realizing subsystems in: xDSL modems, disk drive read channels, biomedical electronics, Bluetooth/ZigBee applications, phaselocked loops, FM stereo demodulator, touch-tone telephone tone decoder and crossover network used in a three-way high-fidelity loudspeaker etc. A number of design methods for companding filters and their building blocks have been introduced in the literature. Most of the proposed filter structures operate either above 1.5V or under symmetrical (1.5V) power supplies. According to data that provides information about the near future of semiconductor technology, International Technology Roadmap for Semiconductors (ITRS), in 2013, the supply voltage of digital circuits in 32 nm technology will be 0.5 V. Therefore, the trend for the implementation of analog integrated circuits is the usage of low-voltage building blocks that use a single 0.5-1.5V power supply. Therefore, the present investigation was primarily concerned with the study and design of low voltage and low power Companding filters. The work includes the study about: the building blocks required in implementing low voltage and low power Companding filters; the techniques used to realize low voltage and low power Companding filters and their various areas of application. Various novel low voltage and low power Companding filter designs have been developed and studied for their characteristics to be applied in a particular portable area of application. The developed designs include the N-th order universal Companding filter designs, which have been reported first time in the open literature. Further, an endeavor has been made to design Companding filters with orthogonal tuning of performance parameters so that the designs can be simultaneously used for various features. The salient features of each of the developed circuit are described. Electronic tunability is one of the major features of all of the designs. Use of grounded capacitors and resistorless designs in all the cases makes the designs suitable for IC technology. All the designs operate in a low-voltage and low-power environment essential for portable system applications. Unless specified otherwise, all the investigations on these designs are based on the PSPICE simulations using model parameters of the NR100N bipolar transistors and BSIM 0.35μm/TSMC 0.25μm /TSMC 0.18μm CMOS process MOS transistors. The performance of each circuit has been validated by comparing the characteristics obtained using simulation with the results present in the open literature. The proposed designs could not be realized in silicon due to non-availability of foundry facility at the place of study. An effort has already been started to realize some of the designs in silicon and check their applicability in practical circuits. At the basic level, one of the proposed Companding filter designs was implemented using the commercially available transistor array ICs (LM3046N) and was found to verify the theoretical predictions obtained from the simulation results

Wavelet Theory

The wavelet is a powerful mathematical tool that plays an important role in science and technology. This book looks at some of the most creative and popular applications of wavelets including biomedical signal processing, image processing, communication signal processing, Internet of Things (IoT), acoustical signal processing, financial market data analysis, energy and power management, and COVID-19 pandemic measurements and calculations. The editor’s personal interest is the application of wavelet transform to identify time domain changes on signals and corresponding frequency components and in improving power amplifier behavior

Spatially distributed computational modeling of a nonlinear vibrating string

Värähtelevän kielen epälineaarinen käyttäytyminen saa monissa kielisoittimissa aikaan soittimelle luonteenomaisen ja helposti tunnistettavan äänen. Laadukkaan kielisoitinsynteesin vuoksi onkin tärkeää, että nykyaikaiset äänisynteesimenetelmät ottavat huomioon myös kielten epälineaarisuudet. Tässä diplomityössä esitellään kaksi uutta synteesimenetelmää, jotka fysikaalisen mallinnuksen avulla simuloivat epälineaarisia näpättyjä kieliä paikkajakautuneesti, keskittyen jännitysmodulaation tuottamiin epälineaarisuuksiin. Toinen menetelmistä käyttää hajautettuja murtoviivesuotimia digitaalisen aaltojohtomallin viivesilmukan pituuden ajonaikaisessa virittämisessä, kun taas toinen hyödyntää murtoviivesuotimia äärelliseen erotukseen pohjautuvan mallin aikaresoluution muuttamisessa ajon aikana. Jännitysmodulaation suuruus arvioidaan kummankin mallin tapauksessa jokaisella aika-askeleella kielen pidentymästä. Molempien mallien simulaatiotulokset esitellään ja niitä verrataan toisiinsa sekä myös mitattuihin arvoihin. Epälineaarisen aaltojohtomallin avulla on toteutettu reaaliaikainen kantelemalli.Nonlinearities in string instruments are responsible for several interesting acoustical features, resulting in characteristic and easily recognizable tones. For this reason, modern synthesis models have to be capable of modeling this nonlinear behavior, when high quality results are desired. This thesis presents two novel physical modeling algorithms for simulating the tension modulation nonlinearity in plucked strings in a spatially distributed manner. The first method uses fractional delay filters within a digital waveguide structure, allowing the length of the string to be modulated during run time. The second method uses a nonlinear finite difference approach, where the string state is approximated between sampling instants also using fractional delay filters, thus allowing run-time modulation of the temporal sampling location. The magnitude of the tension modulation is evaluated from the elongation of the string at every time step in both cases. Simulation results of the two models are presented and compared. Real-time sound synthesis of the kantele, a traditional Finnish plucked-string instrument with strong effect of tension modulation, has been implemented using the nonlinear digital waveguide algorithm

Re-Sonification of Objects, Events, and Environments

abstract: Digital sound synthesis allows the creation of a great variety of sounds. Focusing on interesting or ecologically valid sounds for music, simulation, aesthetics, or other purposes limits the otherwise vast digital audio palette. Tools for creating such sounds vary from arbitrary methods of altering recordings to precise simulations of vibrating objects. In this work, methods of sound synthesis by re-sonification are considered. Re-sonification, herein, refers to the general process of analyzing, possibly transforming, and resynthesizing or reusing recorded sounds in meaningful ways, to convey information. Applied to soundscapes, re-sonification is presented as a means of conveying activity within an environment. Applied to the sounds of objects, this work examines modeling the perception of objects as well as their physical properties and the ability to simulate interactive events with such objects. To create soundscapes to re-sonify geographic environments, a method of automated soundscape design is presented. Using recorded sounds that are classified based on acoustic, social, semantic, and geographic information, this method produces stochastically generated soundscapes to re-sonify selected geographic areas. Drawing on prior knowledge, local sounds and those deemed similar comprise a locale's soundscape. In the context of re-sonifying events, this work examines processes for modeling and estimating the excitations of sounding objects. These include plucking, striking, rubbing, and any interaction that imparts energy into a system, affecting the resultant sound. A method of estimating a linear system's input, constrained to a signal-subspace, is presented and applied toward improving the estimation of percussive excitations for re-sonification. To work toward robust recording-based modeling and re-sonification of objects, new implementations of banded waveguide (BWG) models are proposed for object modeling and sound synthesis. Previous implementations of BWGs use arbitrary model parameters and may produce a range of simulations that do not match digital waveguide or modal models of the same design. Subject to linear excitations, some models proposed here behave identically to other equivalently designed physical models. Under nonlinear interactions, such as bowing, many of the proposed implementations exhibit improvements in the attack characteristics of synthesized sounds.Dissertation/ThesisPh.D. Electrical Engineering 201