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Signal Processing Magazine, IEEE

Issue 2 • Date March 2007

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Displaying Results 1 - 25 of 32
  • Front cover - IEEE Signal Processing Magazine

    Page(s): c1
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  • Contents

    Page(s): 1
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  • Write Feature Articles with a Lasting Impact [From the Editor]

    Page(s): 2
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  • Maintaining our agility [President's Message]

    Page(s): 4
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  • 2007 IEEE Workshop on Machine Learning for Signal Processing [Call for Papers]

    Page(s): 5
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  • 37 SPS Members Elevated to Fellows [Society News]

    Page(s): 6 - 7
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  • Call for nominations

    Page(s): 7
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  • Signal Processing for Sound Synthesis: Computer-Generated Sounds and Music for All [From the Guest Editors]

    Page(s): 8 - 10
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  • Toward Next-Generation Digital Keyboard Instruments

    Page(s): 12 - 20
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    In this article, alternative approaches to digital keyboard instrument synthesis are looked into. Physics-based sound synthesis, which aims at generating natural-sounding musical instrument tones algorithmically without using a large sample database, is a promising approach. It would provide high-quality music synthesis to systems that cannot afford a large memory, such as mobile phones and portable electronic games. The realistic parametric synthesis of musical instrument sounds is still a challenge, but physical modeling techniques introduced during the last few decades can help to solve it. Recently, the first commercial products have been introduced, for example, by Pianoteq. Three keyboard instruments, the clavichord, the harpsichord, and the grand piano, are focused on here. The sound production principles and acoustics of these instruments are first discussed. Then, the previous parametric synthesis algorithms developed for these instruments are reviewed. The remaining part of this article concentrates on new signal processing methods for parametric synthesis of the piano View full abstract»

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  • Wave-based Simulation of Wind Instrument Resonators

    Page(s): 21 - 31
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    This article focuses on modular, wave-based, discrete-time modeling of the resonator oscillations. The main advantage of using a wave-based approach is that it explicitly simulates the way in which waves build up in a real instrument and therefore inherently yields a spatial representation. The wave-based approach also happens to be computationally efficient, mainly due to the fact that lossless traveling of waves in 1-D waveguides can be implemented with very few arithmetic operations View full abstract»

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  • Robust Physical Modeling Sound Synthesis for Nonlinear Systems

    Page(s): 32 - 41
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    Direct numerical simulation methods have played a growing role, albeit a background one, in musical sound synthesis for some time now. Yet, they appear to be the most straightforward and general approach to physical modeling. For such general techniques applied to complex nonlinear systems, the problem of numerical stability lies on the horizon. This article has attempted to outline a more modern time-domain technique for the analysis and construction of physical modeling sound synthesis algorithms, which addresses this issue, as well as some of the more delicate questions of boundary termination. Neither of these is dealt with adequately using frequency domain methods. In essence, the closer attention one pays to the underlying dynamics (particularly the energetic properties), the more robust a method may be constructed View full abstract»

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  • Blocked-based physical modeling for digital sound synthesis

    Page(s): 42 - 54
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    Block-based physical modeling is a methodology for modeling physical systems with different subsystems. It is an important concept for the physical modeling of real or virtual musical instruments where different components may be modeled according to different paradigms. Connecting systems of diverse nature in the discrete-time domain requires a common interconnection strategy. This contribution presents suitable interconnection strategies that incorporate a wide range of modeling blocks and considers the automatic implementation of block-based structures. Software environments are presented, which allow to build complex sound synthesis systems without burdening the user with problems of block compatibility View full abstract»

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  • Acoustic Modeling Using the Digital Waveguide Mesh

    Page(s): 55 - 66
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    The digital waveguide mesh has been an active area of music acoustics research for over ten years. Although founded in 1-D digital waveguide modeling, the principles on which it is based are not new to researchers grounded in numerical simulation, FDTD methods, electromagnetic simulation, etc. This article has attempted to provide a considerable review of how the DWM has been applied to acoustic modeling and sound synthesis problems, including new 2-D object synthesis and an overview of recent research activities in articulatory vocal tract modeling, RIR synthesis, and reverberation simulation. The extensive, although not by any means exhaustive, list of references indicates that though the DWM may have parallels in other disciplines, it still offers something new in the field of acoustic simulation and sound synthesis View full abstract»

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  • Synthesis of the Singing Voice by Performance Sampling and Spectral Models

    Page(s): 67 - 79
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    This paper introduces the concept of synthesis based on performance sampling. It explains that although sampling has been considered a way to capture and reproduce the sound of an instrument, it should be better considered a way to model the sonic space produced by a performer with an instrument. The paper presents a singing voice synthesizer, pointing out the main issues and complexities emerging along its design. Although the current system is able to generate convincing results in certain situations, there is still much room for improvements, especially in the areas of expression, spectral modeling and sonic space design. However, computer singing is definitely coming close to becoming indistinguishable from human performances View full abstract»

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  • Music Synthesis with Reconstructive Phrase Modeling

    Page(s): 80 - 91
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    This article describes a new synthesis technology called reconstructive phrase modeling (RPM). A goal of RPM is to combine the realistic sound quality of sampling with the performance interaction of functional synthesis. Great importance is placed on capturing the dynamics of note transitions-slurs, legato, bow changes, etc. Expressive results are achieved with conventional keyboard controllers. Mastery of special performance techniques is not needed. RPM is an analysis-synthesis system that is related to two important trends in computer music research. The first is a form of additive synthesis in which sounds are represented as a sum of time-varying harmonics plus noise elements. RPM creates expressive performances by searching a database of idiomatic instrumental phrases and combining modified fragments of these phrases to form a new expressive performance. This approach is related to another research trend called concatenative synthesis View full abstract»

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  • Corpus-Based Concatenative Synthesis

    Page(s): 92 - 104
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    Corpus-based concatenative methods for musical sound synthesis have attracted much attention recently. They make use of a variety of sound snippets in a database to assemble a desired sound or phrase according to a target specification given in sound descriptors or by an example sound. With ever-larger sound databases easily available, together with a pertinent description of their contents, they are increasingly used for composition, high-level instrument synthesis, and interactive exploration of a sound corpus. This article gives an overview of the components needed for corpus-based concatenative synthesis and details of some realizations. Signal processing methods are crucial for all parts of analysis, (segmentation; and descriptor analysis), for synthesis, and can intervene in the selection part, e.g., for spectral matching View full abstract»

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  • Fractal additive synthesis

    Page(s): 105 - 115
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    In this article, we outlined time-frequency techniques that, when brought together, form the basis for FAS. The method is based on an exact orthogonal transform, the HBWT. The FAS model can be seen as an intelligent spectrogram, i.e., as a spectrogram where the frequency bins (the HBWT subbands) are adapted to the spectrum of the analyzed sound by tuning the number of channels P to the period of the sound. Local cosines replace the classical windowed complex exponentials, whose main property is that the basis elements form sidebands of the harmonics rather than being centered on the harmonics themselves. The wavelet transform nonuniform frequency subdivision characteristic is exploited to synthesize each sideband of the harmonic peaks by means of colored noise, generating an approximate pseudoperiodic 1/f behavior. The pseudoharmonics are modeled by narrow sidebands whose weights can be generated by means of amplitude envelopes and phase functions in a complexified HBWT domain. We also showed how perceptual criteria can be employed to reduce the number of synthesis parameters. The possibility of independently controlling the noisy components and the harmonic terms and the definition of parameters such as volumes and envelopes for all of the components independently provide powerful tools for processing voiced-sound for sound design purposes View full abstract»

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  • Antialiasing Oscillators in Subtractive Synthesis

    Page(s): 116 - 125
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    In this article, oscillator algorithms for digital subtractive synthesis were reviewed. The algorithms were divided into three categories: bandlimited, quasi-bandlimited, and alias-reducing methods. In the first category, the most interesting methods are in practice those that utilize wavetable techniques. The second category consists of methods that low-pass-filter the underlying continuous-time signal prior to sampling. The optimization of the previously introduced BLIT and BLEP methods were considered as a filter design problem. A new technique called the PolyBLEP method was introduced as a variation of the BLEP method that does not require a table lookup but is based on a closed-form formula. In the PolyBLEP algorithm, an integrated polynomial interpolation function is used for acquiring samples to correct the transition region of the waveform. In the third category, the DPW oscillator algorithm generates an approximate sawtooth waveform that has reduced aliasing. This recently proposed method is probably the simplest useful technique for this purpose, because only the trivial sawtooth is simpler, but it is practically useless due to its heavy aliasing. An alternative decimation filter was proposed for the DPW2X algorithm to suppress aliasing well in the frequency region where human hearing is most sensitive View full abstract»

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  • Concert Hall Acoustics and Audience Perception [Applications Corner]

    Page(s): 126 - 131
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    Concert hall design is known to be a compromise between construction features (such as the architecture/shape of the hall and the number of seats) and sound qualities (such as the clarity, loudness, and reverberation). In this article, the physical properties of unamplified concert halls and their connection to the perception of music and reverberation by humans are discussed View full abstract»

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  • Forensic Automatic Speaker Recognition [Exploratory DSP]

    Page(s): 132 - 135
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    This article discusses the approaches, potential, and limitations of forensic automatic speaker recognition (FASR) which is an exploratory application of digital speech signal processing and pattern recognition for judicial purposes, particularly for law enforcement View full abstract»

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  • A Systems Biology Perspective on Signal Processing in Genetic Network Motifs [Life Sciences]

    Page(s): 136 - 147
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    This article summarizes the characteristics and information processing roles of motifs found in gene transcription networks. The gene transcription networks are defined and the genetic network motif functions are examined. After expanding the discussion to integrated cellular network motifs, the directions for future work are outlined View full abstract»

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  • Computer Music Composition for Children [DSP Education]

    Page(s): 140 - 143
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  • Efficient Multitone Detection [DSP Tips & Tricks]

    Page(s): 144 - 147
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    This article presents the digital signal processing (DSP) tricks employed to build a computationally efficient multitone detection system, implemented without multiplications and with minimal data memory requirements. More specifically, the article describes the detection of incoming dial tones, validity checking to differentiate valid tones from noise signals, and the efficient implementation of the detection system. While the discussion focuses on dual-tone multifrequency (DTMF) telephone dial tone detection, the processing tricks presented may be employed in other multitone detection systems View full abstract»

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  • The H.264/AVC Video Coding Standard [Standards in a Nutshell]

    Page(s): 148 - 153
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  • Eval-Ware: Multimodal Interaction [Best of the Web]

    Page(s): 154 - 155
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Aims & Scope

IEEE Signal Processing Magazine publishes tutorial-style articles on signal processing research and applications, as well as columns and forums on issues of interest.

Full Aims & Scope

Meet Our Editors

Editor-in-Chief
Min Wu
University of Maryland, College Park
United States 

http://www/ece.umd.edu/~minwu/