Audio Signal Processing

A  base-4  leading  zero  detector  (LZD)  design  is  proposed  in  this  paper.  The  design  is  similar  to  the
approach  originally  proposed  by  V.G.  Oklobdzija  with  a  different  technique.  The  circuit  modules  used  in
the  base-4  LZD  approach  are  designed  and  several  N-bit  LZD  circuits  are  implemented  with  a  standard-
cell  realization  in  the  Taiwan  Semiconductor  Manufacturing  Company  (TSMC)  0.65um  CMOS  process.
The  performance  and  layout  area  of  the  base-4  LZD  realization  is  compared  for  implementations  that
contain only 4-to-1 and 2-to-1 multiplexers

Composers working in the sonic arts have frequently found themselves attempting to use spatial audio in ways that didn’t work as intended. Maybe more than any other facet of technological music, mastering spatial audio seems to involve a learning process in which one slowly discovers the things that work and those that don’t. The purpose of this paper is to foster understanding of spatial audio through examples of practical problems. These problems include both some general misconceptions about spatial hearing and some specific examples of things gone wrong. A particular lesson to be learned from this discussion is that there is no silver bullet for solving spatial audio problems, and every situation needs to be understood in appropriate terms.

Filter banks with fixed time-frequency resolution, such as the Short-Time Fourier Transform (STFT), are a common tool for many audio analysis and processing applications allowing effective implementation via the Fast Fourier Transform (FFT). The fixed time-frequency resolution of the STFT can lead to the undesirable smearing of events in both time and frequency. In this paper, we suggest adaptively varying STFT time-frequency resolution in order to reduce filter bank-specific artifacts while retaining adequate frequency resolution. Several strategies for systematic adaptation of time-frequency resolution are proposed. The introduced approach is demonstrated as applied to spectrogram displays, noise reduction, and spectral effects processing.

Mixing multitrack music is an expert task where characteristics of the individual elements and their sum are manipulated in terms of balance, timbre and positioning, to resolve technical issues and to meet the creative vision of the artist or engineer. In this paper we conduct a mixing experiment where eight songs are each mixed by eight different engineers. We consider a range of features describing the dynamic, spatial and spectral characteristics of each track, and perform a multidimensional analysis of variance to assess whether the instrument, song and/or engineer is the determining factor that explains the resulting variance, trend, or consistency in mixing methodology. A number of assumed mixing rules from literature are discussed in the light of this data, and implications regarding the automation of various mixing processes are explored. Part of the data used in this work is published in a new online multitrack dataset through which public domain recordings, mixes, and mix ...

Flamenco singing is characterized by pitch instability, micro-tonal ornamentations, large vibrato ranges, and a high degree of melodic variability. These musical features make the automatic identification of flamenco singers a difficult computational task. In this article we present an end-to-end pipeline for flamenco singer identification based on acoustic motif embeddings. In the approach taken, the fundamental frequency obtained directly from the raw audio signal is approximated. This approximation reduces the high variability of the audio signal and allows for small melodic patterns to be discovered using a sequential pattern mining technique, thus creating a dictionary of motifs. Several acoustic features are then used to extract fixed length embeddings of variable length motifs by using convolutional architectures. We test the quality of the embeddings in a flamenco singer identification task, comparing our approach with previous deep learning architectures, and study the effe...

An approach to watermarking digital signals using frequency modulation-'Chirp Coding'-is considered. The principles underlying this approach are based on the use of a matched filter to reconstruct a 'chirp stream'code that is uniquely robust. The method is generic in the sense that it can, in principle, be used for a variety of different signal (the authentication of speech and biomedical signals, for example). Further, by generating a bit stream that is signal dependent, chirp coding provides a method of self-authentication, ...

This article refers to ongoing research related to the Postdoctoral project " multichannel audio devices applied to music, sound art and bioacoustics " developed by the author for UFF (2016-2018) with support from CAPES and in collaboration with the Laboratory of Acoustics and Sound Arts – LASom/DM/IA/UNICAMP. This work aims to (1) consolidate a spatial sound projection device involving signal processing; and (2) its application in the arts (music) and science (biology and ecology) domains.

These projects deal with my intention to offer the musician novel tools for performance and/or improvisation techniques. I think of them in contrast with traditional modular approach. What I have in mind is neither a system with static parameters, e.g. a step sequencer with fixed rhythmic patterns or divisions and fixed interval ratio, nor a pure stochastic-based system. Moreover, these two projects are related to each other so to create a more general environment that may be used both in musical and in an interactive domain.
The entire work starts from my Bachelor’s thesis, ‘Real-Time Intelligent Harmonizer based on AMDF Pitch Detection’.

Objective quality assessment models have been used more and more in recent years to assess or monitor speech and audio quality in many multime-dia and audio processing systems. These methods offer a clear and repeatable way to evaluate a customer experience by measuring perceived quality on a subjective scale, which is easily understood, such as a quality rating scale, ranging from excellent quality to a low quality. Subsequently, the aim of service providers is to offer reliable services providing the end-user/customer with the best possible quality in the context of the current network conditions to avoid a customer churn. This paper presents a design and performance evaluation of parametric models estimating the audio quality experienced by the end user of broadcasting systems and web-casting applications. The Random Forest (RF) algorithm is used to design non-intrusive parametric models, establishing the relationship between the feature description and the perceived quality scores. For this, the broadcast and web-cast sub-databases were created , where the web-cast sub-database includes 17,280 degraded samples and the broadcast sub-database contains 1,080 degraded samples obtained from the Slovak Radio. The results reported for the proposed paramet-ric audio quality models have validated Random Forest as a powerful technique that provides a good efficiency in terms of Pearson Correlation Coefficient (PCC) and Root Mean Squared Error (RMSE).

Natural Language Processing is in growing demand with recent developments. This Generator model is one such example of a music generation system conditioned on lyrics. The model proposed has been tested on songs having lyrics written only in English, but the idea can be generalized to various languages. This paper’s objective is to mainly explain how one can create a music generator using statistical machine learning methods. This paper also explains how effectively outputs can be formulated, which are the music signals as they are million sized over a short period frame. The parameters mentioned in the paper only serve an explanatory purpose. This paper discusses the effective statistical formulation of output thereby decreasing the vast amount of estimation of output parameters, and how to reconstruct the audio signals from predicted parameters by using ‘phase-shift algorithm

The automated extraction of chord labels from audio recordings constitutes a major task in music information retrieval. To evaluate computer-based chord labeling procedures , one requires ground truth annotations for the underlying audio material. However, the manual generation of such annotations on the basis of audio recordings is tedious and time-consuming. On the other hand, trained musicians can easily derive chord labels from symbolic score data. In this paper, we bridge this gap by describing a procedure that allows for transferring annotations and chord labels from the score domain to the audio domain and vice versa. Using music synchronization techniques, the general idea is to locally warp the annotations of all given data streams onto a common time axis, which then allows for a cross-domain evaluation of the various types of chord labels. As a further contribution of this paper, we extend this principle by introducing a multi-perspective evaluation framework for simultaneously comparing chord recognition results over multiple performances of the same piece of music. The revealed inconsistencies in the results do not only indicate limitations of the employed chord labeling strategies but also deepen the understanding of the underlying music material.

The computer-based harmonic analysis of music recordings with the goal to automatically extract chord labels directly from the given audio data constitutes a major task in music information retrieval. In most automated chord recognition procedures, the given music recording is first converted into a sequence of chroma-based audio features and then pattern matching techniques are applied to map the chroma features to chord labels. In this paper, we analyze the role of the feature extraction step within the recognition pipeline of various chord recognition procedures based on template matching strategies and hidden Markov models. In particular, we report on numerous experiments which show how the various procedures depend on the type of the underlying chroma feature as well as on parameters that control temporal and spectral aspects.

The concept of similarity matrices (SMs) has been widely used for a multitude of music analysis and retrieval tasks including audio structure analysis or version identification. For such tasks, the improvement of structural properties of the similarity matrix at an early state of the processing pipeline has turned out to be of crucial importance. In this paper, we present the SM toolbox, which contains MATLAB implementations for computing and enhancing similarity matrices in various ways. Furthermore, our toolbox includes a number of additional tools for parsing, navigation, and visualization synchronized with audio playback. Finally, we provide the code for a recently proposed audio thumbnailing procedure that demonstrates the applicability and importance of enhancement concepts. Providing MATLAB implementations on a website under a GNU-GPL license and including many illustrative examples, our aim is to foster research and education in music information retrieval.

Humans tend to organize perceived information into hierarchies and structures, a principle that also applies to music. Even musically untrained listeners unconsciously analyze and segment music with regard to various musical aspects, for example, identifying recurrent themes or detecting temporal boundaries between contrasting musical parts. This paper gives an overview of state-of-the-art methods for computational music structure analysis, where the general goal is to divide an audio recording into temporal segments corresponding to musical parts and to group these segments into musically meaningful categories. There are many different criteria for segmenting and structuring music audio. In particular, one can identify three conceptually different approaches, which we refer to as repetition-based, novelty-based, and homogeneity-based approaches. Furthermore, one has to account for different musical dimensions such as melody, harmony, rhythm, and timbre. In our state-of-the-art report, we address these different issues in the context of music structure analysis, while discussing and categorizing the most relevant and recent articles in this field.

As a result of massive digitization efforts and the world wide web, there is an exploding amount of available digital data describing and representing music at various semantic levels and in diverse formats. For example, in the case of the Beatles songs, there are numerous recordings including an increasing number of cover songs and arrangements as well as MIDI data and other symbolic music representations. The general goal of music synchronization is to align the multiple information sources related to a given piece of music. This becomes a difficult problem when the various representations reveal significant differences in structure and polyphony, while exhibiting various types of artifacts. In this paper, we address the issue of how music synchronization techniques are useful for automatically revealing critical passages with significant difference between the two versions to be aligned. Using the corpus of the Beatles songs as test bed, we analyze the kind of differences occurring in audio and MIDI versions available for the songs.

La realizzazione del brano Incontro di Giorgio Nottoli, ci ha posto di fronte a varie e profondamente diverse questioni, esse affrontate con i giusti mezzi ci hanno permesso di iniziare a realizzare gli strumenti necessari per l’interpretazione ed una futura esecuzione dello stesso.
I documenti forniti dall’autore, oltre ad il manuale dello strumento principe(l’EMS VCS3) di questa composizione hanno reso più semplice il lavoro di porting dello strumento, facendoci intendere a fondo come realizzare digitalmente uno strumento analogico complesso e composto da varie tipologie interconnesse di hardware.
FAUST2 e Max3 ci hanno aiutato lungo il percorso; permettendoci di realizzare lo strumento digitale ponendoci una "tabula rasa" capace di farci capire ed intendere quali scelte algoritmiche effettuare a seconda degli aspetti fisici necessari, mostrando ed ampliando le capacità espressive dell’esecutore ed estendendole attraverso l’utilizzo di adeguati mezzi elettronici.
Il seguente testo affronterà dunque uno degli aspetti piú interessanti dello strumento VCS3, ovvero la matrice di cui lo stesso si avvale per connettere tutti i frammenti del "puzzle strumentale elettronico", facendo da fil rouge tra filtri oscillatori ed effetti che Zinovieff4 introdusse in questo primo strumento "commerciale".

The focus of this thesis is on the signal processing techniques used to increase the au-dio quality of the most common digital audio effects employed in electronic musical instruments also taking into account the feasibility of the proposed algorithms’ implementation in accordance with the design constraints and the available computational limits. More in detail four different issues have been analyzed throughout this dissertation: artificial reverberation, analysis and emu-lation of nonlinear devices, audio morphing, and room response equalization.
Among the audio effects, one of the most used is definitely artificial reverberation. A great deal of research has been devoted in the last decades to improve the performance of digital artificial reverberators. Thanks to the progress of technology the traditional techniques based on recursive structures (i.e., IIR filters) are accompanied by new approaches based on fast convolution tech-niques and hybrid reverberator structures. On this basis, an efficient real-time implementation of a fast convolution algorithm has been proposed taking into account an embedded system. More-over, a technique for reducing the computational load required by this operation using psycho-acoustic expedients has been presented considering a joint assessment of the energy decay relief and the absolute threshold of hearing. Finally, some techniques for the approximation of the convolution operation with recursive structures at low computational cost have been suggested.
Although the convolution operation allows the exact reproduction of a linear system, it is im-portant to consider that most of the audio effects are nonlinear systems (i.e., compressors, distor-tion, amplifiers). For this reason, the most commonly used techniques for the emulation of non-linear systems based on a black box approach have been studied and analyzed. In particular, a technique for the approximation of the dynamic convolution operation by exploiting the princi-pal component analysis has been proposed. Using this procedure it is possible to reduce the cost of dynamic convolution without lowering the perceived audio quality. An adaptive algorithm for the identification of nonlinear systems using orthogonal functions has also been presented.
In order to provide greater flexibility and major artistic expression to musicians, several audio morphing techniques have been analyzed. In particular, this procedure makes possible to com-bine two or more audio signals in order to create new sounds that are acoustically interesting. This study has led to the development of an audio morphing algorithm for percussive hybrid sound generation. The main features of the presented approach are preprocessing of the audio references performed in the frequency domain and time domain linear interpolation to execute the morphing.
Finally, equalization techniques for improving the quality of sound reproduction systems by compensating the room transfer function have been taken into account. In particular, two algo-rithms for adaptive minimum-phase equalization and a mixed-phase equalization technique have been proposed. In order to verify the suitability of the proposed systems, experiments on a realis-tic scenario have been carried out.

This paper reviews 3D sound technology. It focuses on the core technical issue of 3D sound: the scientific and engineering means by which a listener of a stereo reproduction system can perceive the direction of a sound in a 3D space. The scientific basis is first discussed, followed by the practical techniques for 3D stereo reproduction.

The decomposition of audio signals into perceptually meaningful modulation components is highly desirable for the development of new audio effects on the one hand and as a building block for future efficient audio compression algorithms on the other hand. In the past, there has always been a distinction between paramet-ric coding methods and waveform coding: While waveform coding methods scale easily up to transparency (provided the necessary bit rate is available), parametric coding schemes are subjected to the limitations of the underlying source models. Otherwise, parametric methods usually offer a wealth of manipu-lation possibilities which can be exploited for application of audio effects, while waveform coding is strictly limited to the best as possible reproduction of the original signal. The analy-sis/synthesis approach presented in this paper is an attempt to show a way to bridge this gap by enabling a seamless transition between both approaches.

Assim como muitas formas de arte surgiram no século XX, mixar músicas caminhou junto com as inovações tecnológicas que envolve som e imagem. Num diálogo leve com o leitor, Reynaldo Leite-expõe o conceito fundamental para você exercitar sua criatividade, sem fórmulas de sucesso ou regras rígidas. Um sólido embasamento teórico e prático, útil para iniciantes e iniciados na arte da mixagem.
Disponível na Amazon Books.

The design and implementation of lossless audio signal processing using Finite Field Transforms is discussed. Finite field signal processing techniques are described. The effects of filter length and coefficient accuracy are also discussed. Finite field transform algorithms which would be suitable for lossless signal processing are presented

One of the challenges in computational acoustics is the identification of models that can simulate and predict the physical behavior of a system generating an acoustic signal. Whenever such models are used for commercial applications an additional constraint is the time-to-market, making automation of the sound design process desirable. In previous works, a computational sound design approach has been proposed for the parameter estimation problem involving timbre matching by deep learning, which was applied to the synthesis of pipe organ tones. In this work we refine previous results by introducing the former approach in a multi-stage algorithm that also adds heuristics and a stochastic optimization method operating on objective cost functions based on psychoacoustics. The optimization method shows to be able to refine the first estimate given by the deep learning approach and substantially improve the objective metrics, with the additional benefit of reducing the sound design process time. Subjective listening tests are also conducted to gather additional insights on the results. Index Terms-physics-based acoustic modeling, neural networks , computational sound design, iterative optimization

RESUMO: O Jazz apresenta uma abundância de elementos musicais/extramusicais, estes cultivados diante de um processo dicótomo, de um povo que experimentou os dissabores/sabores frente aos limites/possibilidades, perseguição/resistência. Essa dualidade de prós vs contras, nos apresenta recursos significativos para o processo de análise/estruturação composicional do gênero. Todavia, no tocante ao estudo de áudio sua abordagem parece ser precária, o que nos leva a pensar que há pouca literatura em língua portuguesa sobre o conhecimento sociocultural, como recurso para auxiliar os engenheiros de áudio. Deste modo, é factível afirmar que estes profissionais podem apropriar-se desse conhecimento como guia imagético do som para uma reprodução fidedigna? Este artigo busca propor uma metodologia de aplicação em engenharia de áudio para auxiliar os engenheiros no processo de captação e uso de equipamentos sem comprometer com a identidade sonora do jazz. Para tal, faz-se necessário realizar uma discussão sobre o contexto sociocultural e apresentar um projeto musical, já concluído, o qual levou em consideração a aplicação conceitual. Diante disto, foi atestado que não basta ser apenas técnico, é necessário ter o conhecimento sobre o assunto em questão, apontando que o conhecimento interdisciplinar pode gerar resultados sonoros com fidedignidade a estética jazzística.

Palavras-chave: Jazz, estética sonora e engenharia de áudio.

This paper presents a low latency pitch shifting algorithm based on the Short-Time Fourier Transform (STFT). Unlike existing STFT-based implementations of pitch shifting, the presented algorithm is more robust to reductions of the Fourier transform size. As a result, it achieves latencies as low as 12ms and still produces good quality, whereas other algorithms are performing much worse with similar low latency constraints. The presented algorithm also provides an alternate way of mitigating the well-known phasiness problem of the phase vocoder.

An audio-assisted system is investigated that detects if a movie scene is a dialogue or not. The system is based on actor indicator functions. That is, functions which define if an actor speaks at a certain time instant. In particular, the cross-correlation and the magnitude of the corresponding the cross-power spectral density of a pair of indicator functions are input to various classifiers, such as voted perceptrons, radial basis function networks, random trees, and support vector machines for dialogue/non-dialogue detection. To boost classifier efficiency AdaBoost is also exploited. The aforementioned classifiers are trained using ground truth indicator functions determined by human annotators for 41 dialogue and another 20 non-dialogue audio instances. For testing, actual indicator functions are derived by applying audio activity detection and actor clustering to audio recordings. 23 instances are randomly chosen among the aforementioned 41 dialogue instances, 17 of which correspond to dialogue scenes and 6 to non-dialogue ones. Accuracy ranging between 0.739 and 0.826 is reported.

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This book is designed to provide comprehensive of Multimedia Technologies with a strong focus on field practice and active experimentation. Amply supplemented with rich pedagogy, it offers an unparalleled learning experience to both students and practitioners.
Available both in eBook and Print format. Concise coverage and provides practical tips for laboratory assignments using MATLAB, Adobe Flash, HTML, and other popular tools. Foreword by Prof. Philip Barker, School of Computing, University of Teesside, Middlesbrough, UK.
ISBN: 9781259083310

Reviews:
Indian Journal of Open Learning, (19(1), 2010: http://journal.ignouonline.ac.in/iojp/index.php/IJOL/article/view/113-
Journal of Educational Technology & Society, 13(2), 2010: http://www.ifets.info/journals/13_2/20.pdf-
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Electronic  communication  is  increasingly  susceptible  to  eavesdropping  and  malicious  interventions.  The  issues  of  security  and  privacy  have  traditionally  been
approached  using  tools  from  cryptography  and steganography.
Steganography  can  be  feasible alternative to cryptography in various countries where usage of encryption is illegal.
In  this  paper,  a  novel  scheme  of  data  hiding  is  introduced  which  provide  high  level  of  security to digital media.4LSB and phase encoding algorithm are used for data embedding in  video and audio files respectively. Quality of video file is strictly preserved even after secret  data  embedding.  Experimental  results  have  demonstrated  the  feasibility  and  efficiency  of  the proposed work.