Download A Robust and Multi‐Scale Modal Analysis This paper presents a new approach to modal synthesis for rendering sounds of virtual objects. We propose a generic method for modal analysis that preserves sound variety across the surface of an object, at different scales of resolution and for a variety of complex geometries. The technique performs automatic voxelization of a surface model and automatic tuning of the parameters of hexahedral finite elements, based on the distribution of material in each cell. The voxelization is performed using a sparse regular grid embedding of the object, which easily permits the construction of plausible lower resolution approximations of the modal model. With our approach, we can compute the audible impulse response of a variety of objects. Our solution is robust and can handle non-manifold geometries that include both volumetric and surface parts, such as those used in games, training simulations, and other interactive virtual environment.
Download Recent CCRMA research in Digital Audio Synthesis, Processing and Effects This extended abstract summarizes DAFx-related developments at CCRMA over the past year or so.
Download Physical Modeling for Spatial Sound Synthesis This contribution combines techniques for sound synthesis and spatial reproduction for a joint synthesis of the sound production and sound propagation properties of virtual string instruments. The generated sound field is reproduced on a massive multichannel loudspeaker system using wave field synthesis techniques. From physical descriptions of string vibrations and sound waves by partial differential equations follows an algorithmic procedure for synthesis-driven wave field reproduction. Its processing steps are derived by mathematical analysis and signal processing principles. Three different building blocks are addressed: The simulation of string vibrations, a model for the radiation pattern of the generated acoustical waves, and the determination of the driving signals for the multichannel loudspeaker array. The proposed method allows the spatial reproduction of synthetic spatial sound without the need for pre-recorded or pre-synthesized source tracks.
Download Transaural Stereo in a Beamforming Approach This paper presents a study on algorithms for headphone-free binaural synthesis using a dedicated loudspeaker configuration. Both algorithms under investigation improve the properties of the binaural synthesis performance of the array. Firstly, beam-forming provides sound radiation localized at two freely adjustable, narrow target spots. Adjusting both spots to the locations of the listener’s ears achieves a good basis. Secondly, an additional interaural crosstalk canceler improves the overall result.
Download Rendering of an acoustic beam through an array of loudspeakers This paper addresses the problem of rendering a virtual source through loudspeaker arrays. The orientation of the virtual source and its aperture determine its radial beampattern. The methodology we present here imposes that the wavefield in a predetermined listening area best approximates the desired wavefield in the least squares sense. With respect to the traditional techniques the number of constraints is much higher than the number of loudspeakers. As a consequence, the loudspeaker coefficient vector is the solution of an over-determined equation system. Moreover this system may be ill-conditioned. In order to solve these issues, we resort to a least squares inversion combined with a Singular Value Decomposition (SVD) to attenuate the problem of ill-conditioning. Some experimental results show the feasibility and the issues of this methodology.
Download Spectral Dealy Filters with Feedback Delay Filters with Feedback and Time-Varying Coefficients A recently introduced structure to implement a continuously smooth spectral delay, based on a cascade of first-order allpass filters and an equalizing filter, is described and the properties of this spectral delay filter are reviewed. A new amplitude envelope equalizing filter for the spectral delay filter is proposed and the properties of structures utilizing feedback and/or time-varying filter coefficients are discussed. In addition, the stability conditions for the feedback and the time-varying structures are derived. A spectral delay filter can be used for synthesizing chirp-like sounds or for modifying the timbre of arbitrary audio signals. Sound examples on the use of the spectral delay filters utilizing the structures discussed in this paper can be found at http://www.acoustics.hut. fi/publications/papers/dafx09-sdf/.
Download SMSPD, LIBSMS and a Real‐Time SMS Instrument We present a real-time implementation of SMS synthesis in Pure Data. This instrument focuses on interaction with the ability to continuously synthesize any frame position within an SMS sound representation, in any order, thereby freeing time from other parameters such as frequency or spectral shape. The instrument can be controlled expressively with a Wacom Tablet that offers both coupled and absolute controls with good precision. A prototype graphical interface in python is presented that helps to interact with the SMS data through visualization. In this system, any sound sample with interesting spectral features turns into a playable instrument. The processing functionality originates in the SMS C code written almost 20 years ago, now re-factored into the open source library, libsms, also wrapped into a python module. A set of externals for Pure Data, called smspd, was made using this library to facilitate on-the-fly analysis, flexible modifications, and interactive synthesis. We discuss new transformations are introduced based on the possibilities of this system and ideas for higher-level, feature based transformations that benefit from the interactivity of this system.
Download Reservoir Computing: a powerful Framework for Nonlinear Audio Processing This paper proposes reservoir computing as a general framework for nonlinear audio processing. Reservoir computing is a novel approach to recurrent neural network training with the advantage of a very simple and linear learning algorithm. It can in theory approximate arbitrary nonlinear dynamical systems with arbitrary precision, has an inherent temporal processing capability and is therefore well suited for many nonlinear audio processing problems. Always when nonlinear relationships are present in the data and time information is crucial, reservoir computing can be applied. Examples from three application areas are presented: nonlinear system identification of a tube amplifier emulator algorithm, nonlinear audio prediction, as necessary in a wireless transmission of audio where dropouts may occur, and automatic melody transcription out of a polyphonic audio stream, as one example from the big field of music information retrieval. Reservoir computing was able to outperform state-of-the-art alternative models in all studied tasks.
Download Automatic Noise Gate Settings for Multitrack Drum Recordings A method has been developed for automating the settings of a noise gate. The method has been applied to a kick drum track containing bleed from secondary drum sources and white noise. The optimal settings are found by maximising the signal to distortion ratio (SDR). The SDR has contributions from the distortion caused to the kick drum signal, and the residual bleed and noise. These two components are weighted, enabling the gate to be controlled by a single parameter. It is shown that the improvement in the SDR can be obtained when the two components of the SDR are approximated, enabling the optimal settings to be calculated from the noisy signal and a single kick drum hit. It is found that the optimal threshold is slightly above the peak level of the noise component of the signal.
Download Blind Separation of Monaural Signals using Complex Wavelets In this paper, a new method of blind source separation of monaural signals is presented. It is based on similarity criteria between envelopes and frequency trajectories of the components of the signal, and on its onset and offset times. The main difference with previous works is that in this paper, the input signal has been filtered using a flexible complex band pass filter bank that is a discrete version of the Complex Continuous Wavelet Transform (CCWT). Our main purpose is to show that the CCWT can be a powerful tool in blind separation, due to its strong coherence in both time and frequency domains. The presented separation algorithm is a first approximation to this important task. An example set of four synthetically mixed monaural signals have been analyzed by this method. The obtained results are promising.