What sensory feedback, tactile or auditory, is the more important for a musician when playing? In an attempt to answer this question, subjects were asked to play along with a metronome while the auditory feedback from their playing was manipulated. The preliminary results showed a tendency for matching sound with sound, i.e. players initiated strokes earlier as the delay increased. Increase in timing errors indicate a possible breakpoint around 55 ms. As the feedback was delayed even more subjects showed increased difficulties in maintaining a steady rhythm.

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Digital waveguide mesh models have provided an accurate and efficient method of modelling the properties of many resonant structures, including acoustic spaces. 2-D rectilinear and triangular mesh structures have been used extensively in the past to model plates and membranes and as potential analogues to 2-D acoustic spaces. This paper looks at current developments relating to this technique and attempts to highlight potential ways forward for this research. This includes an investigation into the potential suitability of this technique for surround-sound applications and a realtime implementation as a VST plugin.

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This paper describes the prototype of a system for recognition of hummed tunes. The prototype takes as input an unknown hummed tune, turns it into segments representing the distinct notes of the tune, detects the pitch of the segments and turns them into a simple note-like symbolic representation to be matched against a reference database of known tunes. The system returns a list of tunes sorted according to similarity to the hummed tune. Each entry in the list contains information like title, composer, author, and a link to a midi-file enabling the user to ensure himself that the he is reading the data of the song that he has hummed. Testing with seven persons humming 46 songs in total, has shown a successful recognition rate of 75.6% with a search space of 39925 songs.

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In many applications of audio signal processing modeling of the signal is required. The most commonly used approach for audio signal modeling is to assume the audio signal as an (autoregressive) AR-process where the audio signal is locally stationary over a relatively short time interval. In this case the audio signal can be modeled with an all-pole IIR (infinite impulse response) filter, which leads to LPC (linear predictive coding) where the current input sample is predicted by a linear combination of past samples of the input signal. However, in practice the relatively short time interval (i.e. a frame) where the signal is stationary will vary significantly in the audio signal data stream. Also the information content of the frames will show considerable variation. For a proper modeling of an audio signal it is essential that a suitable frame size and appropriate number of model parameters is used instead of a constant frame size and model order. In this paper we present an adaptive frame-by-frame technique for modeling audio signals, which automatically adjusts the optimal modeling frame size and the optimal number of model parameters for each frame.

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With the huge increase in the availability of digital music, it has become more important to automate the task of querying a database of musical pieces. At the same time, a computational solution of this task might give us an insight into how humans perceive and classify music. In this paper, we discuss our attempts to classify music into three broad categories: rock, classical and jazz. We discuss the feature extraction process and the particular choice of features that we used- spectrograms and mel scaled cepstral coefficients (MFCC). We use the texture-of- texture models to generate feature vectors out of these. Together, these features are capable of capturing the frequency-power profile of the sound as the song proceeds. Finally, we attempt to classify the generated data using a variety of classifiers. we discuss our results and the inferences that can be drawn from them.

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Synthetic reverberation is generally derived by using filter-based algorithms to mimic the types of impulse responses which are generated in natural room responses. Standard approaches would include Schroeder parallel comb filter, feedback delay networks and nested all-pass structures. One of the difficulties in specifying a useable reverb algorithm is that sonic quality is sensitive to the parameter choices used. For example, inappropriate choice of gain and delay in a Schroeder reverberator leads to what is referred to as ‘graineyness’ or ‘fluttering’. These are subjective descriptions of perceived aural characteristics associated with reverb response. Because of the qualitative nature of these artifacts, parameter specification for any new reverb structure generally involves subjective assessment of sonic quality - a ‘tweak-and-listen’ approach. This paper attempts to take some initial steps toward objective assessment of synthetic reverberation by identifying possible quantitative measures which correlate with these perceptual features of reverberation response. These quantitative measures are derived from a novel joint time frequency distribution which incorporates accurate ear masking effects. This has been developed by generating ear-response based smoothing kernels for the Wigner Distribution leading to its designation as the EarWig Distribution (EWD). Thus, the EWD of an audio signal provides us with a representation which highlights perceptually relevant signal features only. From the EWD of reverb responses it is shown that it is possible to identify some objective signal characteristics which contribute to perceived graineyness and fluttering. This provides the basis for the establishment of associated quantitative measures which would be a significant contribution in synthetic reverberation design.

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This paper presents current work being made in the development of an ergonomic graphical environment for MPEG-4 SA instruments creation and manipulation. The project goal is to define a means of giving easy access to the power and flexibility of the MPEG-4 SA framework.

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XML is a subset of SGML (ISO 8879), the international standard metalanguage for text markup systems. It has been widely used in data representation and interchange formats. This research proposes to investigate the use of XML for musical score representation.

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In this article, we propose a new Query-by-humming Music Retrieval System, focusing on the nature of hummed queries, and more precisely, on their non-tempered characteristics. We show that avoiding pitch query quantization gives better retrieval results. The study of frequential imprecision of hummed melodies also allows us to present a new and easy way of stimulating systems for their quality evaluation.

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The control of directivity represents a new stake in the reproduction of sound sources by an electroacoustic device. This paper first reminds the principle of a general method for reproducing the acoustical field radiated by a source. The present study, is devoted to the bandwidth improvement and to the optimization of the directivity control. It rests on the preliminary constitution of a set of basic directivities from which it is possible to tune the cardioicity and the threedimensional orientation of the synthesized directivity pattern. For the simulation of musical instruments, it is important to maintain independent control on the spatial and spectral characteristics. For that, an analysis/synthesis method of the power spectrum is used to provide the automatic design of a correction filter to applied on near field recording before feeding the source. Results are discussed considering room acoustics criteria and examples of musical instruments.

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