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

Volume diffraction gratings for optical telecommunications applications: design study for a spectral equalizer

The main characteristics required for a diffraction grating used for demultiplexing functions in spectral equalizing systems are investigated, both theoretically and experimentally. We show that volume-phase holographic (VPH) gratings can be used as dispersive elements instead of classic reflection surface-relief gratings presently employed in most optical telecommunications devices. A design method for this type of diffraction grating and experimental results are presented, confirming that VPH gratings are well suited to such applications

Design Considerations for a Digital Audio Equalizer

The objective of this thesis is to consider a method for designing a digital audio equalizer. The primary design criteria is minimum audible frequency response error between a digital and a reference analog equalizer throughout the entire audio frequency range from 20 Hz to 20 Khz. The first step is to obtain a set of analog filters that suitably represent the reference equalization. From these filters, digital filter coefficients are generated using the bilinear transformation. Then, the digital filters are combined with anti-aliasing and D/A reconstruction filters and a zero-order hold to complete the design. Analysis of methods to minimize frequency axis warping effects on the response of the high frequency filters is presented. The problems associated with realizing a filter with low natural frequency and a very high sample rate is also studied

Dynamic link budget simulation

A new simulator named DLBS (Dynamic Link Budget Simulator) was written to simulate the time-varying communication link between a vehicle that re-enters the atmosphere from the outer space, and a ground station. During the vehicle descent trajectory, communications blackouts typically occur due to the effects of plasma that forms around the vehicle. A companion simulator, AIPT (Antenna In Plasma Tool), evaluates the electric field at the input of the ground station antenna, taking into consideration the vehicle structure, its antenna, the characteristics of plasma at some specified points along the vehicle trajectory, and the obtained values are stored in a file. DLBS processes the data read from the AIPT output file and evaluates the corresponding channel transfer functions. DLBS then allows to simulate the typical telemetry and telecommand links, using both CCSDS standardised and some non standard channel encoding schemes and modulations. For each generated frame, DLBS uses a channel transfer function obtained by adequately interpolating the two nearest transfer functions evaluated in the initial phase. DLBS includes realistic frame, frequency, phase and bit synchronisation, so that synchronisation errors are also included as source of performance degradation, and measures both the average bit and frame error rates, and the bit error rate at frame level, so that it is possible to appreciate the dynamic system behaviour. The paper will show the results obtained for a case stud

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

Ground Robotic Hand Applications for the Space Program study (GRASP)

This document reports on a NASA-STDP effort to address research interests of the NASA Kennedy Space Center (KSC) through a study entitled, Ground Robotic-Hand Applications for the Space Program (GRASP). The primary objective of the GRASP study was to identify beneficial applications of specialized end-effectors and robotic hand devices for automating any ground operations which are performed at the Kennedy Space Center. Thus, operations for expendable vehicles, the Space Shuttle and its components, and all payloads were included in the study. Typical benefits of automating operations, or augmenting human operators performing physical tasks, include: reduced costs; enhanced safety and reliability; and reduced processing turnaround time

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

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

Analog Computing using 1T1R Crossbar Arrays

Memristor is a novel passive electronic device and a promising candidate for new generation non-volatile memory and analog computing. Analog computing based on memristors has been explored in this study. Due to the lack of commercial electrical testing instruments for those emerging devices and crossbar arrays, we have designed and built testing circuits to implement analog and parallel computing operations. With the setup developed in this study, we have successfully demonstrated image processing functions utilizing large memristor crossbar arrays. We further designed and experimentally demonstrated the first memristor based field programmable analog array (FPAA), which was successfully configured for audio equalizer and frequency classifier demonstration as exemplary applications of such memristive FPAA (memFPAA)