Download Fluently Remixing Musical Objects with Higher-Order Functions Soon after the Echo Nest Remix API was made publicly available and open source, the primary author began aggressively enhancing the Python framework for re-editing music based on perceptually-based musical analyses. The basic principles of this API – integrating content-based metadata with the underlying signal – are described in the paper, then the authors’ enhancements are described. The libraries moved from supporting an imperative coding style to incorporating influences from functional programming and domain specific languages to allow for a much more fluent, terse coding style, allowing users to concentrate on the functions needed to find the portions of the song that were interesting, and modifying them. The paper then goes on to describe enhancements involving mixing multiple sources with one another and enabling user-created and user-modifiable effects that are controlled by direct manipulation of the objects that represent the sound. Revelations that the Remix API does not need to be as integrated as it currently is point to future directions for the API at the end of the paper.
Download Handling Inharmonic Series with Median-Adjustive Trajectories A new method for the analysis of inharmonic instrumental tones is presented. The method exploits an equation derived from the well-know inharmonic series equation, where the inharmonicity coefficient is balanced with the frequencies and numbers of any two partials extracted from a pseudo-harmonic series. A serial search for increasingly deviating spectral peaks is aided with the integrated refinement of increasingly reliable inharmonicity coefficient and fundamental frequency estimates. This firsthand approach to the problem of evaluating inharmonic spectra brings about an unprecedented level of simplicity, efficiency and accuracy.
Download KRONOS ‐ A Vectorizing Compiler for Music DSP This paper introduces Kronos, a vectorizing Just in Time compiler designed for musical programming systems. Its purpose is to translate abstract mathematical expressions into high performance computer code. Musical programming system design criteria are considered and a three-tier model of abstraction is presented. The low level expression Metalanguage used in Kronos is described, along with the design choices that facilitate powerful, yet transparent vectorization of the machine code.
Download Compositional Sketches in PWGLSynth PWGLSynth has already a long history in controlling physicsbased instruments. The control system has been score-based, i.e. the user prepares a score in advance, and by interactive listening process the result can be be refined either by adjusting score information, performance rules and/or the visual instrument definition. This scheme allows detailed control on how the instrument model reacts to control information generated from the score. This paper presents a complementary approach to sound synthesis where the idea is to generate algorithmically typically relatively short musical textures. The user can improvise with various compositional ideas, adjust parameters, and listen to the results in real-time either individually or interleaved. This is achieved by utilizing a special code-box scheme that allows any textual Lisp expression to be interfaced to the visual part of the PWGL system.
Download Tools for Interactive Audio Signal Analysis based on Sliding DFT This article describes an application the author developed in order to compare analysis and synthesis of musical audio signals through Short Time Fourier transform (STFT), Constant Q and Sliding Discrete Fourier Transform (SDFT). This software is the basis for applications of SDFT and Constant Q to other consolidated synthesis techniques. By itself, it is a stand alone instrument for calculating and quickly comparing spectrum analysis and synthesis of musical signals. No expertise is required and it can for example be easily used by music composers without in depth knowledge of DSP processing tools.
Download A VST Reverberation Effect Plugin Based on Synthetic Room Impulse Responses In this paper we present a newly developed VST reverberation effect plugin (“HybridReverb”) based on synthetic room impulse responses (RIRs). We detail how we choose proper parameters for the synthesis of RIRs as presets for our convolution-based reverberation effect. The implemented stereo/surround plugin provides natural sounding reverberation based on physical principles. The newly developed convolution engine features signal processing with low latency and uniform processing load.
Download A spectral subtraction rule for real‐time DSP implementation of noise reduction in speech signals Spectral subtraction is a method for restoration of the spectrum magnitude for signals observed in additive noise, through subtraction of an estimate of the average noise spectrum from the noisy signal spectrum. In this paper we show that, starting from the known minimum mean-square error (MMSE) suppression rules of Ephraim and Malah and under the same modeling assumptions, a simpler suppression filtering rule can be found. Moreover, we demonstrate its performances and compare its computational costs with respect to the reference rule of Ephraim and Malah. This result permits a real time implementation of the exposed theory with an efficient algorithm on the DSP TMS320 C6713B.
Download Two‐Dimensional Fourier Processing of Rasterised Audio There is continuous research effort into the expansion and refinement of transform techniques for audio signal processing needs, yet the two-dimensional Fourier transform has seldom been applied to audio. This is probably because audio does not readily allow the application of a 2D transform, unlike images for which its use is common. A signal mapping is first required to obtain a two-dimensional representation. However the 2D Fourier transform opens up potential for new or improved analysis and transformation of audio. In this paper, raster scanning is used to provide a simple mapping between one- and two-dimensional representations. This allows initial experimentation with the 2D Fourier transform, in which the 2D spectrum can be observed. A straightforward method is used to display the spectral data as a colour image. The transform gives information on two frequency axes, one in the typical audible frequency range and the other in the low frequency rhythmic range. This representation can be used to more easily observe rhythmic modulations in the signal. Some novel audio transformations are presented, allowing manipulation of rhythmic frequency content. The techniques developed using the 2D Fourier transform allow interaction with audio in a new domain, both analytically and creatively. This work shows how two common signal processing mechanisms can be combined to exciting effect for audio applications.
Download Performance of source spatialization and source localization Algorithms using Conjoint Models of Interaural Level and Time Cues In this paper, we describe a head-model based on interaural cues (e.g. interaural level differences and interaural time differences). Based on this model, we proposed, in previous works, a binaural source spatialization method (SSPA), that we extended to a multispeaker spatialization technique that works on a speaker array in a pairwise motion (MSPA) [1], [2]. Here, we evaluate the spatialization techniques, and compare them to well-known methods (e.g. VBAP (Vector Base Amplitude Panning) [3]). We also test the robustness of a adapted conjoint localization method under noisy and reverberant conditions; this method uses spectra of recorded binaural signals, and tries to minimize the distance between the ILD and ITD based azimuth estimates. We show comparative results with the PHAT generalized cross-correlation localization method [4].