Efficient target-response interpolation for a graphic equalizer

Proceedings of the 41st IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP, held in Shanghai (China) during 20-25 March 2016.A graphic equalizer is an adjustable filter in which the command gain of each frequency band is practically independent of the gains of other bands. Designing a graphic equalizer with a high precision requires evaluating a target response that interpolates the magnitude response at several frequency points between the command gains. Good accuracy has been previously achieved by using polynomial interpolation methods such as cubic Hermite or spline interpolation. However, these methods require large computational resources, which is a limitation in real-time applications. This paper proposes an efficient way of computing the target response without sacrificing the approximation accuracy. This new approach called Linear Interpolation with Constant Segments (LICS) reduces the computing time of the target response by 55% and has an intrinsic parallel structure. Performance of the LICS method is assessed on an ARM Cortex-A7 core, which is commonly used in embedded systems.This work was conducted in spring 2015 when the first author was a visiting postdoctoral researcher at Aalto University. This research has been partly funded by the TIN2014-53495-R and TIN2011-23283 projects of the Ministerio de Economía y Competitividad and FEDER

Efficient Algorithms for Immersive Audio Rendering Enhancement

Il rendering audio immersivo è il processo di creazione di un’esperienza sonora coinvolgente e realistica nello spazio 3D. Nei sistemi audio immersivi, le funzioni di trasferimento relative alla testa (head-related transfer functions, HRTFs) vengono utilizzate per la sintesi binaurale in cuffia poiché esprimono il modo in cui gli esseri umani localizzano una sorgente sonora. Possono essere introdotti algoritmi di interpolazione delle HRTF per ridurre il numero di punti di misura e per creare un movimento del suono affidabile. La riproduzione binaurale può essere eseguita anche dagli altoparlanti. Tuttavia, il coinvolgimento di due o più gli altoparlanti causa il problema del crosstalk. In questo caso, algoritmi di cancellazione del crosstalk (CTC) sono necessari per eliminare i segnali di interferenza indesiderati. In questa tesi, partendo da un'analisi comparativa di metodi di misura delle HRTF, viene proposto un sistema di rendering binaurale basato sull'interpolazione delle HRTF per applicazioni in tempo reale. Il metodo proposto mostra buone prestazioni rispetto a una tecnica di riferimento. L'algoritmo di interpolazione è anche applicato al rendering audio immersivo tramite altoparlanti, aggiungendo un algoritmo di cancellazione del crosstalk fisso, che considera l'ascoltatore in una posizione fissa. Inoltre, un sistema di cancellazione crosstalk adattivo, che include il tracciamento della testa dell'ascoltatore, è analizzato e implementato in tempo reale. Il CTC adattivo implementa una struttura in sottobande e risultati sperimentali dimostrano che un maggiore numero di bande migliora le prestazioni in termini di errore totale e tasso di convergenza. Il sistema di riproduzione e le caratteristiche dell'ambiente di ascolto possono influenzare le prestazioni a causa della loro risposta in frequenza non ideale. L'equalizzazione viene utilizzata per livellare le varie parti dello spettro di frequenze che compongono un segnale audio al fine di ottenere le caratteristiche sonore desiderate. L'equalizzazione può essere manuale, come nel caso dell'equalizzazione grafica, dove il guadagno di ogni banda di frequenza può essere modificato dall'utente, o automatica, la curva di equalizzazione è calcolata automaticamente dopo la misurazione della risposta impulsiva della stanza. L'equalizzazione della risposta ambientale può essere applicata anche ai sistemi multicanale, che utilizzano due o più altoparlanti e la zona di equalizzazione può essere ampliata misurando le risposte impulsive in diversi punti della zona di ascolto. In questa tesi, GEQ efficienti e un sistema adattativo di equalizzazione d'ambiente. In particolare, sono proposti e approfonditi tre equalizzatori grafici a basso costo computazionale e a fase lineare e quasi lineare. Gli esperimenti confermano l'efficacia degli equalizzatori proposti in termini di accuratezza, complessità computazionale e latenza. Successivamente, una struttura adattativa in sottobande è introdotta per lo sviluppo di un sistema di equalizzazione d'ambiente multicanale. I risultati sperimentali verificano l'efficienza dell'approccio in sottobande rispetto al caso a banda singola. Infine, viene presentata una rete crossover a fase lineare per sistemi multicanale, mostrando ottimi risultati in termini di risposta in ampiezza, bande di transizione, risposta polare e risposta in fase. I sistemi di controllo attivo del rumore (ANC) possono essere progettati per ridurre gli effetti dell'inquinamento acustico e possono essere utilizzati contemporaneamente a un sistema audio immersivo. L'ANC funziona creando un'onda sonora in opposizione di fase rispetto all'onda sonora in arrivo. Il livello sonoro complessivo viene così ridotto grazie all'interferenza distruttiva. Infine, questa tesi presenta un sistema ANC utilizzato per la riduzione del rumore. L’approccio proposto implementa una stima online del percorso secondario e si basa su filtri adattativi in sottobande applicati alla stima del percorso primario che mirano a migliorare le prestazioni dell’intero sistema. La struttura proposta garantisce un tasso di convergenza migliore rispetto all'algoritmo di riferimento.Immersive audio rendering is the process of creating an engaging and realistic sound experience in 3D space. In immersive audio systems, the head-related transfer functions (HRTFs) are used for binaural synthesis over headphones since they express how humans localize a sound source. HRTF interpolation algorithms can be introduced for reducing the number of measurement points and creating a reliable sound movement. Binaural reproduction can be also performed by loudspeakers. However, the involvement of two or more loudspeakers causes the problem of crosstalk. In this case, crosstalk cancellation (CTC) algorithms are needed to delete unwanted interference signals. In this thesis, starting from a comparative analysis of HRTF measurement techniques, a binaural rendering system based on HRTF interpolation is proposed and evaluated for real-time applications. The proposed method shows good performance in comparison with a reference technique. The interpolation algorithm is also applied for immersive audio rendering over loudspeakers, by adding a fixed crosstalk cancellation algorithm, which assumes that the listener is in a fixed position. In addition, an adaptive crosstalk cancellation system, which includes the tracking of the listener's head, is analyzed and a real-time implementation is presented. The adaptive CTC implements a subband structure and experimental results prove that a higher number of bands improves the performance in terms of total error and convergence rate. The reproduction system and the characteristics of the listening room may affect the performance due to their non-ideal frequency response. Audio equalization is used to adjust the balance of different audio frequencies in order to achieve desired sound characteristics. The equalization can be manual, such as in the case of graphic equalization, where the gain of each frequency band can be modified by the user, or automatic, where the equalization curve is automatically calculated after the room impulse response measurement. The room response equalization can be also applied to multichannel systems, which employ two or more loudspeakers, and the equalization zone can be enlarged by measuring the impulse responses in different points of the listening zone. In this thesis, efficient graphic equalizers (GEQs), and an adaptive room response equalization system are presented. In particular, three low-complexity linear- and quasi-linear-phase graphic equalizers are proposed and deeply examined. Experiments confirm the effectiveness of the proposed GEQs in terms of accuracy, computational complexity, and latency. Successively, a subband adaptive structure is introduced for the development of a multichannel and multiple positions room response equalizer. Experimental results verify the effectiveness of the subband approach in comparison with the single-band case. Finally, a linear-phase crossover network is presented for multichannel systems, showing great results in terms of magnitude flatness, cutoff rates, polar diagram, and phase response. Active noise control (ANC) systems can be designed to reduce the effects of noise pollution and can be used simultaneously with an immersive audio system. The ANC works by creating a sound wave that has an opposite phase with respect to the sound wave of the unwanted noise. The additional sound wave creates destructive interference, which reduces the overall sound level. Finally, this thesis presents an ANC system used for noise reduction. The proposed approach implements an online secondary path estimation and is based on cross-update adaptive filters applied to the primary path estimation that aim at improving the performance of the whole system. The proposed structure allows for a better convergence rate in comparison with a reference algorithm

Solutions for New Terrestrial Broadcasting Systems Offering Simultaneously Stationary and Mobile Services

221 p.[EN]Since the first broadcasted TV signal was transmitted in the early decades of the past century, the television broadcasting industry has experienced a series of dramatic changes. Most recently, following the evolution from analogue to digital systems, the digital dividend has become one of the main concerns of the broadcasting industry. In fact, there are many international spectrum authorities reclaiming part of the broadcasting spectrum to satisfy the growing demand of other services, such as broadband wireless services, arguing that the TV services are not very spectrum-efficient. Apart from that, it must be taken into account that, even if up to now the mobile broadcasting has not been considered a major requirement, this will probably change in the near future. In fact, it is expected that the global mobile data traffic will increase 11-fold between 2014 and 2018, and what is more, over two thirds of the data traffic will be video stream by the end of that period. Therefore, the capability to receive HD services anywhere with a mobile device is going to be a mandatory requirement for any new generation broadcasting system. The main objective of this work is to present several technical solutions that answer to these challenges. In particular, the main questions to be solved are the spectrum efficiency issue and the increasing user expectations of receiving high quality mobile services. In other words, the main objective is to provide technical solutions for an efficient and flexible usage of the terrestrial broadcasting spectrum for both stationary and mobile services. The first contributions of this scientific work are closely related to the study of the mobile broadcast reception. Firstly, a comprehensive mathematical analysis of the OFDM signal behaviour over time-varying channels is presented. In order to maximize the channel capacity in mobile environments, channel estimation and equalization are studied in depth. First, the most implemented equalization solutions in time-varying scenarios are analyzed, and then, based on these existing techniques, a new equalization algorithm is proposed for enhancing the receivers’ performance. An alternative solution for improving the efficiency under mobile channel conditions is treating the Inter Carrier Interference as another noise source. Specifically, after analyzing the ICI impact and the existing solutions for reducing the ICI penalty, a new approach based on the robustness of FEC codes is presented. This new approach employs one dimensional algorithms at the receiver and entrusts the ICI removing task to the robust forward error correction codes. Finally, another major contribution of this work is the presentation of the Layer Division Multiplexing (LDM) as a spectrum-efficient and flexible solution for offering stationary and mobile services simultaneously. The comprehensive theoretical study developed here verifies the improved spectrum efficiency, whereas the included practical validation confirms the feasibility of the system and presents it as a very promising multiplexing technique, which will surely be a strong candidate for the next generation broadcasting services.[ES]Desde el comienzo de la transmisión de las primeras señales de televisión a principios del siglo pasado, la radiodifusión digital ha evolucionado gracias a una serie de cambios relevantes. Recientemente, como consecuencia directa de la digitalización del servicio, el dividendo digital se ha convertido en uno de los caballos de batalla de la industria de la radiodifusión. De hecho, no son pocos los consorcios internacionales que abogan por asignar parte del espectro de radiodifusión a otros servicios como, por ejemplo, la telefonía móvil, argumentado la poca eficiencia espectral de la tecnología de radiodifusión actual. Asimismo, se debe tener en cuenta que a pesar de que los servicios móviles no se han considerado fundamentales en el pasado, esta tendencia probablemente variará en el futuro cercano. De hecho, se espera que el tráfico derivado de servicios móviles se multiplique por once entre los años 2014 y 2018; y lo que es más importante, se pronostica que dos tercios del tráfico móvil sea video streaming para finales de ese periodo. Por lo tanto, la posibilidad de ofrecer servicios de alta definición en dispositivos móviles es un requisito fundamental para los sistemas de radiodifusión de nueva generación. El principal objetivo de este trabajo es presentar soluciones técnicas que den respuesta a los retos planteados anteriormente. En particular, las principales cuestiones a resolver son la ineficiencia espectral y el incremento de usuarios que demandan mayor calidad en los contenidos para dispositivos móviles. En pocas palabras, el principal objetivo de este trabajo se basa en ofrecer una solución más eficiente y flexible para la transmisión simultánea de servicios fijos y móviles. La primera contribución relevante de este trabajo está relacionada con la recepción de la señal de televisión en movimiento. En primer lugar, se presenta un completo análisis matemático del comportamiento de la señal OFDM en canales variantes con el tiempo. A continuación, con la intención de maximizar la capacidad del canal, se estudian en profundidad los algoritmos de estimación y ecualización. Posteriormente, se analizan los algoritmos de ecualización más implementados, y por último, basándose en estas técnicas, se propone un nuevo algoritmo de ecualización para aumentar el rendimiento de los receptores en tales condiciones. Del mismo modo, se plantea un nuevo enfoque para mejorar la eficiencia de los servicios móviles basado en tratar la interferencia entre portadoras como una fuente de ruido. Concretamente, tras analizar el impacto del ICI en los receptores actuales, se sugiere delegar el trabajo de corrección de dichas distorsiones en códigos FEC muy robustos. Finalmente, la última contribución importante de este trabajo es la presentación de la tecnología LDM como una manera más eficiente y flexible para la transmisión simultánea de servicios fijos y móviles. El análisis teórico presentado confirma el incremento en la eficiencia espectral, mientras que el estudio práctico valida la posible implementación del sistema y presenta la tecnología LDM c

Solving Weighted Least Squares (WLS) problems on ARM-based architectures

TheWeighted Least Squares algorithm (WLS) is applied to numerous optimization problems, but requires the use of high computational resources, especially when complex arithmetic is involved. This work aims to accelerate the resolution of a WLS problem by reducing the computational cost (relaying on BLAS/LAPACK routines) and the computational precision from double to single. As a test case, we design an IIR filter for a Graphic Equalizer, where the numerical errors due to single precision are easily visualized. In addition, given the importance of low power architectures for this kind of implementations, we evaluate the performance, scalability, and energy efficiency of each method on two different processors implementing the ARMv7 architecture, widely used in current mobile devices with power constraints. Results show that the method that exhibits a high theoretical computational cost overcomes in efficiency other methods with lower theoretical cost in architectures of this type.This work started in spring 2016 when Jose A. Belloch was a visiting postdoctoral researcher at Budapest University of Technology and Economics thanks to the European Network COST Action IC1305 inside the program Short Term Scientific Mission with the following reference: COST-SPASM-ECOST-STSM-IC1305-020416-072431. Dr. Jose A. Belloch is supported by GVA contract APOSTD/2016/069. The researchers from Universitat Jaume I are supported by the CICYT projects TIN2014-53495-R of MINECO and FEDER. The authors from the Universitat Politecnica de Valencia are supported by MINECO Projects TEC2015-67387-C4-1-R, PROMETEOII/2014/003 and CAPAP-H5 network TIN2014-53522-REDT. The researcher from UCM is supported by the EU (FEDER) and the Spanish MINECO, under Grants TIN 2015-65277-R and TIN2012-32180. The work of Balazs Bank was supported by the UNKP-16-4-III New National Excellence Program of the Ministry of Human Capacities, Hungary.Belloch Rodríguez, JA.; Bank, B.; Igual Peña, FD.; Quintana Ortí, ES.; Vidal Maciá, AM. (2017). Solving Weighted Least Squares (WLS) problems on ARM-based architectures. Journal of Supercomputing. 73(1):530-542. https://doi.org/10.1007/s11227-016-1910-9S530542731Smith TM, van de Geijn RA, Smelyanskiy M, Hammond JR, Van Zee FG (2014) Anatomy of high-performance many-threaded matrix multiplication. In: 28th IEEE International Parallel and Distributed Processing Symposium (IPDPS 2014)Burrus CS (2012) Iterative reweighted least squares. OpenStax-CNC document, May 2012, module m45285. http://cnx.org/content/m45285/1.12 . Accessed 2 Nov 2016Khang SW (1972) Best $$L_p$$ L p approximation. Math Comput 26(118):505–508Jackson LB (2008) Frequency-domain Steiglitz-McBride method for least-squares filter design, ARMA modeling, and periodogram smoothing. IEEE Signal Process Lett 15:49–52Bank B (2012) Magnitude-priority filter design for audio applications. In: Proceedings of $$132^{{\rm nd}}$$ 132 nd AES Convention, Preprint No. 8591, Budapest, Hungary, May 2012Daubechies I, Devire R, Fornasier M, Gntrk CS (2010) Iteratively reweighted least squares minimization for sparse recovery. Comput Music J 23(2):52–69Rämö J, Välimäki V, Bank B (2014) High-precision parallel graphic equalizer. IEEE/ACM Trans Audio Speech Lange Proc 22(12):1894–1904Perez Gonzales E, Reiss J (2009) Automatic equalization of multi-channel audio using cross-adaptive methods. In: Proceedings of AES 127th Convention, New York, Oct. 2009Rämö J, Välimäki V (2013) Live sound equalization and attenuation with a headset. In: Proceedings of AES 51st International Conference, Helsinki, Finland, Aug. 2013Mäkivirta A, Antsalo P, Karjalainen M, Välimäki V (2003) Modal equalization of loudspeaker-room responses at low frequencies. J Audio Eng Soc 51(5):324–343Holters M, Zölzer U (2006) Graphic equalizer design using higher-order recursive filters. In: Proceedings of International Conference Digital Audio Effects, Montreal, QC, pp 37–40Tassart S (2013) Graphical equalization using interpolated filter banks. J Audio Eng Soc 61(5):263–279Chen Z, Geng GS, Yin FL, Hao J (2014) A pre-distortion based design method for digital audio graphic equalizer. Digital Signal Process 25:296–302Välimäki V, Reiss J (2016) All about audio equalization: solutions and frontiers. Appl Sci 6(5):129–145Belloch JA, Välimäki V (2016) Efficient target-response interpolation for a graphic equalizer. In: 2016 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), March 2016, pp 564–568Belloch JA, Alventosa FJ, Alonso P, Quintana-Ortí ES, Vidal AM (2016) Accelerating multi-channel filtering of audio signal on arm processors. J Supercomput, pp 1–12. doi: 10.1007/s11227-016-1689-8Belloch JA, Gonzalez A, Igual FD, Mayo R, Quintana-Ortí ES (2015)Vectorization of binaural sound virtualization on the ARM cortex-A15 architecture. In: Proceedings of 23rd European Signal Processing Conference, (EUSIPCO), Nize, France, September 2015Mitra G, Johnston B, Rendell A, McCreath E, Zhou J (2013) Use of simd vector operations to accelerate application code performance on low-powered arm and intel platforms. In: IEEE 27th International Parallel and Distributed Processing Symposium Workshops PhD Forum (IPDPSW), May 2013, pp 1107–1116Tomov S, Dongarra J, Baboulin M (2008) Towards dense linear algebra for hybrid gpu accelerated manycore systems. LAPACK Working Note, Tech. Rep. 210, Oct. 2008. http://www.netlib.org/lapack/lawnspdf/lawn210.pdf . Accessed 2 Nov 2016Dongarra JJ, DuCroz J, Hammarling S, Hanson RJ (1985) A proposal for an extended set of fortran basic linear algebra subprograms. ACM Signum Newsletter, New York, pp 2–18Golub GH, Loan CFV (2013) Matrix Comput, 4th edn. The John Hopkins University Press, BaltimoreAlonso P, Badia RM, Labarta J, Barreda M, Dolz MF, Mayo R, Quintana-Ortí ES, Reyes R (2012) Tools for power-energy modelling and analysis of parallel scientific applications. In: 41st International Conference on Parallel Processing—ICPP, 2012, pp 420–42

Graafinen ekvalisointi taajuusvarpattujen digitaalisten suotimien avulla

The aim of this thesis is to design a graphic equalizer with frequency warped digital filters. The proposed design consists of a warped FIR filter for the low frequency bands and a standard FIR filter for the high frequency bands. This de- sign is used to implement both an octave and a one-third octave equalizer in Matlab. Low frequency equalization with FIR filters requires high filter orders. The frequency resolution of the lowest band of the graphic equalizer requires filter orders that are impractical for real life applications. With frequency warping filter orders can be lowered, so that a practical graphic equalizer can be designed. With this design common gain build-up problems, which are present in most of the IIR designs, can be avoided. The proposed equalizer design is found to be accurate and comparable to the previous equalizer designs. Filter orders required are small enough to this design to be used in real life applications. The gain build-up problem is avoided in this design, as several equalizer bands are filtered with a single filter. The computational costs of the design are higher than the costs of the other compared designs. However, the difference can be smaller if the accuracy restrictions are lowered.Tämän työn tavoitteena on suunnitella graafinen ekvalisaattori taajuusvarpattujen digitaalisten suotimien avulla. Ehdotettu ekvalisaattorimalli koostuu taajuusvarpatusta ja tavallisesta FIR suotimesta. Varpattua suodinta käytetään alimpien taajuuskaistojen suodattamiseen ja tavallista FIR suodinta ylimpien kaistojen suodattamiseen. Tätä mallia käytetään sekä oktaavi- että terssikaista-ekvalisaattorien totetutamiseen Matlabilla. Matalien taajuuksien ekvalisointi edellyttää korkeaa astelukua FIR suotimilta. Alimpien taajuuskaistojen taajuusresoluutio edellyttää astelukuja, jotka ovat epäkäytännöllisiä tosielämän sovelluksissa. Taajuusvarppauksella suotimien astelukuja voidaan pienentää, jolloin graafinen ekvalisaattori voidaan toteuttaa käytännössä. Tällä mallilla voidaan välttää IIR ekvalisaattorien yleinen ongelma, jossa ekvalisaattorien kaistojen vahvistus vaikuttaa viereisiin kaistoihin. Ehdotettu ekvalisaattorimalli todetaan olevan tarkka ja vertailukelpoinen aikaisempien toteutuksien kanssa. Suotimien asteluvut ovat tarpeeksi pieniä, jotta tätä mallia voidaan käyttää tosielämän toteutuksissa. Kaistojen välinen vaikutus vältetään tällä mallilla, sillä useampi kaista suodatetaan yhdellä suotimella. Laskennallinen kuorma on tällä toteutuksella suurempi kuin muilla vertailluilla toteutuksilla. Eroa voidaan pienentää, jos ekvalisaattorin tarkkuusvaatimuksia lasketaan

Investigation into digital audio equaliser systems and the effects of arithmetic and transform errors on performance

Merged with duplicate record 10026.1/2685 on 07.20.2017 by CS (TIS)Discrete-time audio equalisers introduce a variety of undesirable artefacts into audio mixing systems, namely, distortions caused by finite wordlength constraints, frequency response distortion due to coefficient calculation and signal disturbances that arise from real-time coefficient update. An understanding of these artefacts is important in the design of computationally affordable, good quality equalisers. A detailed investigation into these artefacts using various forms of arithmetic, filter frequency response, input excitation and sampling frequencies is described in this thesis. Novel coefficient calculation techniques, based on the matched z-transform (MZT) were developed to minimise filter response distortion and computation for on-line implementation. It was found that MZT-based filter responses can approximate more closely to s-plane filters, than BZTbased filters, with an affordable increase in computation load. Frequency response distortions and prewarping/correction schemes at higher sampling frequencies (96 and 192 kHz) were also assessed. An environment for emulating fractional quantisation in fixed and floating point arithmetic was developed. Various key filter topologies were emulated in fixed and floating point arithmetic using various input stimuli and frequency responses. The work provides detailed objective information and an understanding of the behaviour of key topologies in fixed and floating point arithmetic and the effects of input excitation and sampling frequency. Signal disturbance behaviour in key filter topologies during coefficient update was investigated through the implementation of various coefficient update scenarios. Input stimuli and specific frequency response changes that produce worst-case disturbances were identified, providing an analytical understanding of disturbance behaviour in various topologies. Existing parameter and coefficient interpolation algorithms were implemented and assessed under fihite wordlength arithmetic. The disturbance behaviour of various topologies at higher sampling frequencies was examined. The work contributes to the understanding of artefacts in audio equaliser implementation. The study of artefacts at the sampling frequencies of 48,96 and 192 kHz has implications in the assessment of equaliser performance at higher sampling frequencies.Allen & Heath Limite

Advanced automatic mixing tools for music

PhDThis thesis presents research on several independent systems that when combined together can generate an automatic sound mix out of an unknown set of multi‐channel inputs. The research explores the possibility of reproducing the mixing decisions of a skilled audio engineer with minimal or no human interaction. The research is restricted to non‐time varying mixes for large room acoustics. This research has applications in dynamic sound music concerts, remote mixing, recording and postproduction as well as live mixing for interactive scenes. Currently, automated mixers are capable of saving a set of static mix scenes that can be loaded for later use, but they lack the ability to adapt to a different room or to a different set of inputs. In other words, they lack the ability to automatically make mixing decisions. The automatic mixer research depicted here distinguishes between the engineering mixing and the subjective mixing contributions. This research aims to automate the technical tasks related to audio mixing while freeing the audio engineer to perform the fine‐tuning involved in generating an aesthetically‐pleasing sound mix. Although the system mainly deals with the technical constraints involved in generating an audio mix, the developed system takes advantage of common practices performed by sound engineers whenever possible. The system also makes use of inter‐dependent channel information for controlling signal processing tasks while aiming to maintain system stability at all times. A working implementation of the system is described and subjective evaluation between a human mix and the automatic mix is used to measure the success of the automatic mixing tools

Design Techniques for High Performance Wireline Communication and Security Systems

As the amount of data traffic grows exponentially on the internet, towards thousands of exabytes by 2020, high performance and high efficiency communication and security solutions are constantly in high demand, calling for innovative solutions. Within server communication dominates todays network data transfer, outweighing between-server and server-to-user data transfer by an order of magnitude. Solutions for within-server communication tend to be very wideband, i.e. on the order of tens of gigahertz, equalizers are widely deployed to provide extended bandwidth at reasonable cost. However, using equalizers typically costs the available signal-to-noise ratio (SNR) at the receiver side. What is worse is that the SNR available at the channel becomes worse as data rate increases, making it harder to meet the tight constraint on error rate, delay, and power consumption. In this thesis, two equalization solutions that address optimal equalizer implementations are discussed. One is a low-power high-speed maximum likelihood sequence detection (MLSD) that achieves record energy efficiency, below 10 pico-Joule per bit. The other one is a phase-shaping equalizer design that suppresses inter-symbol interference at almost zero cost of SNR. The growing amount of communication use also challenges the design of security subsystems, and the emerging need for post-quantum security adds to the difficulties. Most of currently deployed cryptographic primitives rely on the hardness of discrete logarithms that could potentially be solved efficiently with a powerful enough quantum computer. Efficient post-quantum encryption solutions have become of substantial value. In this thesis a fast and efficient lattice encryption application-specific integrated circuit is presented that surpasses the energy efficiency of embedded processors by 4 orders of magnitude.PHDElectrical EngineeringUniversity of Michigan, Horace H. Rackham School of Graduate Studieshttps://deepblue.lib.umich.edu/bitstream/2027.42/146092/1/shisong_1.pd

Architecture and algorithms for the implementation of digital wireless receivers in FPGA and ASIC: ISDB-T and DVB-S2 cases

[EN] The first generation of Terrestrial Digital Television(DTV) has been in service for over a decade. In 2013, several countries have already completed the transition from Analog to Digital TV Broadcasting, most of which in Europe. In South America, after several studies and trials, Brazil adopted the Japanese standard with some innovations. Japan and Brazil started Digital Terrestrial Television Broadcasting (DTTB) services in December 2003 and December 2007 respectively, using Integrated Services Digital Broadcasting - Terrestrial (ISDB-T), also known as ARIB STD-B31. In June 2005 the Committee for the Information Technology Area (CATI) of Brazilian Ministry of Science and Technology and Innovation MCTI approved the incorporation of the IC-Brazil Program, in the National Program for Microelectronics (PNM) . The main goals of IC-Brazil are the formal qualification of IC designers, support to the creation of semiconductors companies focused on projects of ICs within Brazil, and the attraction of semiconductors companies focused on the design and development of ICs in Brazil. The work presented in this thesis originated from the unique momentum created by the combination of the birth of Digital Television in Brazil and the creation of the IC-Brazil Program by the Brazilian government. Without this combination it would not have been possible to make these kind of projects in Brazil. These projects have been a long and costly journey, albeit scientifically and technologically worthy, towards a Brazilian DTV state-of-the-art low complexity Integrated Circuit, with good economy scale perspectives, due to the fact that at the beginning of this project ISDB-T standard was not adopted by several countries like DVB-T. During the development of the ISDB-T receiver proposed in this thesis, it was realized that due to the continental dimensions of Brazil, the DTTB would not be enough to cover the entire country with open DTV signal, specially for the case of remote localizations far from the high urban density regions. Then, Eldorado Research Institute and Idea! Electronic Systems, foresaw that, in a near future, there would be an open distribution system for high definition DTV over satellite, in Brazil. Based on that, it was decided by Eldorado Research Institute, that would be necessary to create a new ASIC for broadcast satellite reception. At that time DVB-S2 standard was the strongest candidate for that, and this assumption still stands nowadays. Therefore, it was decided to apply to a new round of resources funding from the MCTI - that was granted - in order to start the new project. This thesis discusses in details the Architecture and Algorithms proposed for the implementation of a low complexity Intermediate Frequency(IF) ISDB-T Receiver on Application Specific Integrated Circuit (ASIC) CMOS. The Architecture proposed here is highly based on the COordinate Rotation Digital Computer (CORDIC) Algorithm, that is a simple and efficient algorithm suitable for VLSI implementations. The receiver copes with the impairments inherent to wireless channels transmission and the receiver crystals. The thesis also discusses the Methodology adopted and presents the implementation results. The receiver performance is presented and compared to those obtained by means of simulations. Furthermore, the thesis also presents the Architecture and Algorithms for a DVB-S2 receiver targeting its ASIC implementation. However, unlike the ISDB-T receiver, only preliminary ASIC implementation results are introduced. This was mainly done in order to have an early estimation of die area to prove that the project in ASIC is economically viable, as well as to verify possible bugs in early stage. As in the case of ISDB-T receiver, this receiver is highly based on CORDIC algorithm and it was prototyped in FPGA. The Methodology used for the second receiver is derived from that used for the ISDB-T receiver, with minor additions given the project characteristics.[ES] La primera generación de Televisión Digital Terrestre(DTV) ha estado en servicio por más de una década. En 2013, varios países completaron la transición de transmisión analógica a televisión digital, la mayoría de ellas en Europa. En América del Sur, después de varios estudios y ensayos, Brasil adoptó el estándar japonés con algunas innovaciones. Japón y Brasil comenzaron a prestar el servicio de Difusión de Televisión Digital Terrestre (DTTB) en diciembre de 2003 y diciembre de 2007 respectivamente, utilizando Radiodifusión Digital de Servicios Integrados Terrestres (ISDB-T), también conocida como ARIB STD-B31. En junio de 2005, el Comité del Área de Tecnología de la Información (CATI) del Ministerio de Ciencia, Tecnología e Innovación de Brasil - MCTI aprobó la incorporación del Programa CI-Brasil, en el Programa Nacional de Microelectrónica (PNM). Los principales objetivos de la CI-Brasil son la formación de diseñadores de CIs, apoyar la creación de empresas de semiconductores enfocadas en proyectos de circuitos integrados dentro de Brasil, y la atracción de empresas de semiconductores interesadas en el diseño y desarrollo de circuitos integrados. El trabajo presentado en esta tesis se originó en el impulso único creado por la combinación del nacimiento de la televisión digital en Brasil y la creación del Programa de CI-Brasil por el gobierno brasileño. Sin esta combinación no hubiera sido posible realizar este tipo de proyectos en Brasil. Estos proyectos han sido un trayecto largo y costoso, aunque meritorio desde el punto de vista científico y tecnológico, hacia un Circuito Integrado brasileño de punta y de baja complejidad para DTV, con buenas perspectivas de economía de escala debido al hecho que al inicio de este proyecto, el estándar ISDB-T no fue adoptado por varios países como DVB-T. Durante el desarrollo del receptor ISDB-T propuesto en esta tesis, se observó que debido a las dimensiones continentales de Brasil, la DTTB no sería suficiente para cubrir todo el país con la señal de televisión digital abierta, especialmente para el caso de localizaciones remotas, apartadas de las regiones de alta densidad urbana. En ese momento, el Instituto de Investigación Eldorado e Idea! Sistemas Electrónicos, previeron que en un futuro cercano habría un sistema de distribución abierto para DTV de alta definición por satélite en Brasil. Con base en eso, el Instituto de Investigación Eldorado decidió que sería necesario crear un nuevo ASIC para la recepción de radiodifusión por satélite, basada el estándar DVB-S2. En esta tesis se analiza en detalle la Arquitectura y algoritmos propuestos para la implementación de un receptor ISDB-T de baja complejidad y frecuencia intermedia (IF) en un Circuito Integrado de Aplicación Específica (ASIC) CMOS. La arquitectura aquí propuesta se basa fuertemente en el algoritmo Computadora Digital para Rotación de Coordenadas (CORDIC), el cual es un algoritmo simple, eficiente y adecuado para implementaciones VLSI. El receptor hace frente a las deficiencias inherentes a las transmisiones por canales inalámbricos y los cristales del receptor. La tesis también analiza la metodología adoptada y presenta los resultados de la implementación. Por otro lado, la tesis también presenta la arquitectura y los algoritmos para un receptor DVB-S2 dirigido a la implementación en ASIC. Sin embargo, a diferencia del receptor ISDB-T, se introducen sólo los resultados preliminares de implementación en ASIC. Esto se hizo principalmente con el fin de tener una estimación temprana del área del die para demostrar que el proyecto en ASIC es económicamente viable, así como para verificar posibles errores en etapa temprana. Como en el caso de receptor ISDB-T, este receptor se basa fuertemente en el algoritmo CORDIC y fue un prototipado en FPGA. La metodología utilizada para el segundo receptor se deriva de la utilizada para el re[CA] La primera generació de Televisió Digital Terrestre (TDT) ha estat en servici durant més d'una dècada. En 2013, diversos països ja van completar la transició de la radiodifusió de televisió analògica a la digital, i la majoria van ser a Europa. A Amèrica del Sud, després de diversos estudis i assajos, Brasil va adoptar l'estàndard japonés amb algunes innovacions. Japó i Brasil van començar els servicis de Radiodifusió de Televisió Terrestre Digital (DTTB) al desembre de 2003 i al desembre de 2007, respectivament, utilitzant la Radiodifusió Digital amb Servicis Integrats de (ISDB-T), coneguda com a ARIB STD-B31. Al juny de 2005, el Comité de l'Àrea de Tecnologia de la Informació (CATI) del Ministeri de Ciència i Tecnologia i Innovació del Brasil (MCTI) va aprovar la incorporació del programa CI Brasil al Programa Nacional de Microelectrònica (PNM). Els principals objectius de CI Brasil són la qualificació formal dels dissenyadors de circuits integrats, el suport a la creació d'empreses de semiconductors centrades en projectes de circuits integrats dins del Brasil i l'atracció d'empreses de semiconductors centrades en el disseny i desenvolupament de circuits integrats. El treball presentat en esta tesi es va originar en l'impuls únic creat per la combinació del naixement de la televisió digital al Brasil i la creació del programa Brasil CI pel govern brasiler. Sense esta combinació no hauria estat possible realitzar este tipus de projectes a Brasil. Estos projectes han suposat un viatge llarg i costós, tot i que digne científicament i tecnològica, cap a un circuit integrat punter de baixa complexitat per a la TDT brasilera, amb bones perspectives d'economia d'escala perquè a l'inici d'este projecte l'estàndard ISDB-T no va ser adoptat per diversos països, com el DVB-T. Durant el desenvolupament del receptor de ISDB-T proposat en esta tesi, va resultar que, a causa de les dimensions continentals de Brasil, la DTTB no seria suficient per cobrir tot el país amb el senyal de TDT oberta, especialment pel que fa a les localitzacions remotes allunyades de les regions d'alta densitat urbana.. En este moment, l'Institut de Recerca Eldorado i Idea! Sistemes Electrònics van preveure que, en un futur pròxim, no hi hauria a Brasil un sistema de distribució oberta de TDT d'alta definició a través de satèl¿lit. D'acord amb això, l'Institut de Recerca Eldorado va decidir que seria necessari crear un nou ASIC per a la recepció de radiodifusió per satèl¿lit. basat en l'estàndard DVB-S2. En esta tesi s'analitza en detall l'arquitectura i els algorismes proposats per l'execució d'un receptor ISDB-T de Freqüència Intermèdia (FI) de baixa complexitat sobre CMOS de Circuit Integrat d'Aplicacions Específiques (ASIC). L'arquitectura ací proposada es basa molt en l'algorisme de l'Ordinador Digital de Rotació de Coordenades (CORDIC), que és un algorisme simple i eficient adequat per implementacions VLSI. El receptor fa front a les deficiències inherents a la transmissió de canals sense fil i els cristalls del receptor. Esta tesi també analitza la metodologia adoptada i presenta els resultats de l'execució. Es presenta el rendiment del receptor i es compara amb els obtinguts per mitjà de simulacions. D'altra banda, esta tesi també presenta l'arquitectura i els algorismes d'un receptor de DVB-S2 de cara a la seua implementació en ASIC. No obstant això, a diferència del receptor ISDB-T, només s'introdueixen resultats preliminars d'implementació en ASIC. Això es va fer principalment amb la finalitat de tenir una estimació primerenca de la zona de dau per demostrar que el projecte en ASIC és econòmicament viable, així com per verificar possibles errors en l'etapa primerenca. Com en el cas del receptor ISDB-T, este receptor es basa molt en l'algorisme CORDIC i va ser un prototip de FPGA. La metodologia utilitzada per al segon receptor es deriva de la utilitzada per al receptor IRodrigues De Lima, E. (2016). Architecture and algorithms for the implementation of digital wireless receivers in FPGA and ASIC: ISDB-T and DVB-S2 cases [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/61967TESI

Automatic Calibration and Equalization of a Line Array System

This final project presents an automated Public Address processing unit, using delay and magnitude frequency response adjustment. The aim is to achieve a flat frequency response and delay adjustment between different physically-placed speakers at the measuring point, which is nowadays usually made manually by the sound technician. The adjustment is obtained using four signal processing operations to the audio signal: time delay adjustment, crossover filtering, gain adjustment, and graphic equalization. The automation is in the calculation of different parameter sets: estimation of the time delay, the selection of a suitable crossover frequency, and calculation of the gains for a third-octave graphic equalizer. These automatic methods reduce time and effort in the calibration of line-array PA systems, since only three sine sweeps must be played though the sound system. For verifying the functioning of the system, both simulated signals and measurements have been conducted. A 1:10 scale model of a line array system has been designed and constructed in an anechoic chamber to test the automatic calibration and equalization methods and the results are analyzed