182 research outputs found
'Computing modified Bessel functions with large modulation index for sound synthesis applications
Ordinary Bessel functions are a common function used when examining the spectral properties of frequency modulated signals, particularly in sound synthesis applications. Recently, it was shown that modified Bessel functions can also be used for sound synthesis. However, to limit the impact of aliasing distortion when using these functions, it is essential to set an upper limit on the frequency-dependent modulation index used when computing these functions. However, it can be impossible to do this beyond a certain threshold when using standard mathematical software tools such as Matlab, or the scientific toolbox of the Python language, because of numerical overflow issues. This short paper presents an approach to overcome this limitation using the MaxStar algorithm. Results are also presented to demonstrate the usefulness of this solution
A Graph-Theoretic Approach to Timbre Matching
This paper presents a novel approach to the matching problem
associated with timbre morphing. In particular a graph-theoretic technique, that of subgraph isomorphism, is applied to find correspondences between graph representations of a feature set for each timbre. The features are identified from Wigner distributions of the sounds using an adaptation of the McAulay-Quatieri sinusoidal analysis techniques. These features are then
interpreted as nodes in graph representations of timbre. An audio morphing application can then be implemented by the application of spatial warping and linear interpolation of the Wigner distributions based on the correspondences
established
Instantaneous Frequency Approaches for Speech and Audio Signal Analysis
This paper investigates the use of Instantaneous Frequency Distributions for the analysis of Speech and Audio signals. In particular, methods are proposed for fundamental frequency determination of speech and audio, and method for the tracking of resonances, or formants in the case of speech, present in a signal is described. The techniques considered that are applied to the Instantaneous Frequency Distribution of the signal in order to extract a desired feature or features specifically involve the use of dynamic programming in order to produce smooth estimates
Streaming Frequency-domain DAFX in Csound5
This article discusses the implementation of frequency domain
digital audio effects using the Csound 5 music programming
language, with its streaming frequency-domain signal (fsig)
framework. Introduced to Csound 4.13, by Richard Dobson,
it was further extended by Victor Lazzarini in version 5. The
latest release of Csound incorporates a variety of new opcodes
for different types of spectral manipulations. This article
introduces the fsig framework and the analysis and resynthesis
unit generators. It describes in detail the different types of spectral DAFx made possible by these new opcodes
Subgraph Isomorphism applied to feature correspondence in timbre morphing
This paper presents a subgraph isomorphism technique for correspondence matching in a Wigner distributiontimbre morphing procedure. This aim is to overcome the shortcomings found with previous STFT based methods. The results presented demonstrate that this technique produces a perceptually convincing sound
A modified FM synthesis approach to bandlimited signal generation
Techniques for the generation of bandlimited signals for application to digital implementations of subtractive synthesis have been researched by a number of authors. This paper hopes to contribute to the variety of approaches by proposing a technique based on Frequency Modulation (FM) synthesis. This paper presents and explains the equations required for bandlimited pulse generation using modified FM synthesis. It then investigates the relationships
between the modulation index and the quality of the
reproduction in terms of authenticity and aliasing for a sawtooth wave. To determine the performance of this technique in comparison to others two sets of simulation results are offered: the first computes the relative power of the non-harmonic components, and the second uses the Perceptual Evaluation of Audio Quality (PEAQ) algorithm. It is shown that this technique compares well with the alternatives. The paper concludes with suggestions
for the direction of future improvements to the method
A streaming object-oriented implementation of the modal distribution
The Modal distribution is a time-frequency distribution
specifically designed to model the quasi-harmonic, multi-sinusoidal, nature of music signals and belongs to the Cohen general class of time-frequency distributions. A streaming, object-oriented implementation of the Modal distribution is presented which forms the basis for designing other members of the Cohen class. Implementation of this routine in the C++ Sound Object Library provides a fully portable tool for time-frequency analysis across multiple platforms. The theoretical background to the Cohen general class is outlined followed by an explanation of the design and implementation of the Modal distribution in the SndObj library. Suggestions for future extensions to the new
Modal class and its integration with the entire library
are explored
Time-stretching using the instantaneous frequency distribution and partial tracking
This article presents a method of signal timescale modification using spectral analysis-resynthesis. It discusses an alternative technique for instantaneous frequency estimation, the Instantaneous Frequency Distribution (IFD). The partial tracking analysis employed in this process is explained in some detail, followed by a look into the resynthesis method. The article discusses this technique of time-stretching in comparison to the standard phase vocoder process. Performance details and specific aspects of this implementation are examined, including the C++ code for a time-stretching application
Implementing loudness models in Matlab
In the field of psychoacoustic analysis the goal is to construct a transformation that will map a time waveform into a domain that best captures the response of a human perceiving sound. A key element of such transformations is the mapping between the sound intensity in decibels and its actual perceived loudness. A number of different loudness models exist to achieve this mapping. This paper examines implementation strategies for some of the more wellknown
models in the Matlab software environment
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