Note separation in music signal processing becomes difficult when there are overlapping partials from co-existing notes produced by either the same or different musical instruments. In order to deal with this problem, it is necessary to involve certain invariant features of musical instrument sounds into the separation processing. For example, the timbre of a note of a musical instrument may be used as one possible invariant feature. In this paper, a timbre estimate is used to represent this feature such that it becomes a constraint when note separation is performed on a mixture signal. To demonstrate the proposed method, a timedependent recursive regularization analysis is employed. Spectral envelopes of different notes are estimated and a modified parameter update strategy is applied to the recursive regularization process. The experiment results show that the flaws due to the overlapping partial problem can be effectively reduced through the proposed approach.

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In this study, receiver-based audio error concealment in the context of low-latency Audio over IP transmission is analyzed. Therefore, the well-known technique of audio extrapolation is investigated concerning its usability in real-time scenarios, its applied prediction techniques and various transmission parameters. A large-scale automated evaluation with PEAQ and a MUSHRA listening test reveal the performance of the various extrapolation setups. The results show the suitability of extrapolation to perform audio error concealment in real-time and the qualitative superiority of block based methods over sample based methods.

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This paper introduces Kronos VST, an audio effect plugin conforming to the VST 3 standard that can be programmed on the fly by the user, allowing entire signal processors to be defined in real time. A brief survey of existing programmable plugins or development aids for audio effect plugins is given. Kronos VST includes a functional just in time compiler that produces high performance native machine code from high level source code. The features of the Kronos programming language are briefly covered, followed by the special considerations of integrating user programs into the VST infrastructure. Finally, introductory example programs are provided.

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In recent years there has been an increasing amount of interest in the style of synthesis implemented by Don Buchla in his instrument designs from the early 1960s until the present. A key part of the Buchla synthesizer and its characteristic quality is the ’lowpass gate’ filter and the acoustic-like plucked sounds that it provides. In this work we examine the circuit of the low-pass gate, both its audio and control portions. We propose a number of digital models of these circuits, as well as a model of the photoresistive optoisolator or ’vactrol’ used within them. In the case of the audio path of the device, we pay particular attention to maintaining desirable behavior under time-variation of its parameters. The resulting digital model retains much of the interesting character of the analog system, and is computationally cheap enough to use within a standard computer-music setup.

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We present a fast and robust method for approximating sound propagation in situations where audio and video frame rates may differ significantly and positions of sound sources and listeners are only known at discrete times, so that numerically stable velocities are not available. Typical applications include 3D scenes in virtual environments where positions of sources and listeners are determined in real time by user interaction. Our method employs a computationally inexpensive heuristic that converges to the exact solution for constant speeds and achieves convincing Doppler shifts in general.

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The Csoundo library combines the sound compiler Csound with Processing, a java-based programming language for visual applets. The combination of the two provides a framework for the rapid development of interactive audio-visual applications. Csound and Processing have recently been ported to the Android platform so these languages may now be used to develop applications directly for Android devices. However, at present, Csound for Android has only been implemented with basic Android Graphical User Interface(GUI) elements while Processing for Android lacks support for external libraries, including its established audio libraries. Csoundo for Android bridges the gap between the two languages and allows developers to rapidly build interactive audio applications without having to be concerned about the inner-workings of Android development.

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The present paper deals with a framework destined to manage different aspects of the creation of digital audio devices. By means of a domain-specific language modelling aspects like signal processing and user interaction are implemented. The problem of different hardware interfaces is resolved by the definition of a hardware abstraction layer. This layer provides different types of variables and functions. A compiler translates the model referring the functions and variables defined at the hardware abstraction layer. Furthermore, the compiler is able to split the model into different parts that can be run on different hardware components. The communication needed to manage the distributed model is defined and formalized by the framework. A simple example is presented to help explain the framework’s parts, as are the compiler and the execution unit.

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faust2android is a tool that turns a FAUST program into an Android application. Signal processing tasks as well as accessing the audio record and playback resources are done natively in C++ using the Android Native Development Toolkit (NDK). User interface and other components of the application are programmed in JAVA. The implementation as well as issues related to real-time signal processing on Android platforms are discussed. faust2android is part of a larger project whose goal is to build a full FAUST environment for Android: FAUST D ROID.

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Textual music programming languages offer greater expressive power than diagrammatic visual programming languages and semi-modular graphical user interfaces. However, textual music programming languages don’t allow fine-grained incremental updates to the signal flow graph—instead, they only allow course-grained updates at the statement level. In both the diagrammatic visual programming language and the graphical user interface paradigms, users can directly adjust a parameter by editing the value inline, and such an adjustment does not affect the state of any other part of the signal flow graph. For example, adjusting the attack time of an envelope does not affect the contents of a delay line. By contrast, users of textual music programming languages must either: 1) assign names to nodes and then later use a separate statement to adjust the parameter value, or 2) lose node states (eg. envelope positions, instantaneous LFO phases, and delay line contents) by reevaluating the entire original statement or program. We present a new paradigm in which users can directly edit programs without losing state. In our approach, time-varying programs evaluate to time-varying signal processing graphs, and incremental updates to the program result in incremental updates to the signal processing graph.

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In this paper, we present our evaluations in using blind single channel dereverberation on music signals. The target material is heavily reverberated and dynamic range compressed polyphonic music from several genres. The applied dereverberation method is based on spectral subtraction regulated by a time-frequency domain linear predictive model. We present our results on enhancing music signal quality and automatic beat tracking accuracy with the proposed dereverberation method. Signal quality enhancement, measured by improvement in signal to distortion ratio, is achieved for both reverberant and dynamic range compressed signals. Moreover, the algorithm shows potential as a preprocessing method for music beat tracking.

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