Download Sparse and Structured Decompositions of Audio Signals in Overcomplete Spaces We investigate the notion of “sparse decompositions” of audio signals in overcomplete spaces, ie when the number of basis functions is greater than the number of signal samples. We show that, with a low degree of overcompleteness (typically 2 or 3 times), it is possible to get good approximation of the signal that are sparse, provided that some “structural” information is taken into account, ie the localization of significant coefficients that appears to form clusters. This is illustrated with decompositions on a union of local cosines (MDCT) and discrete wavelets (DWT), that are shown to perform well on percussive signals, a class of signals that is difficult to sparsely represent on pure (local) Fourier bases. Finally, the obtained clusters of individuals atoms are shown to carry higher levels of information, such as a parametrization of partials or attacks, and this is potentially useful in an information retrieval context.
Download Detection of Clicks Using Sinusoidal Modeling for the Confirmation of the Clicks This article presents methods for clicks detection in degraded audio recordings. It begins with a brief description of the method implemented in first instance for the detection of clicks in audio sources based on linear prediction. Looking for an improvement of the results obtained with this method, we propose a method based on sinusoidal modeling for the confirmation of the clicks. This method discards clicks that were wrongly detected. This allows the detection of clicks of small amplitude avoiding wrong detections. The results obtained by this method are shown, confirming the good operation. Finally, the method implemented for detection of clicks in naturally degraded audio sources is presented.
Download Audio Analysis, Visualization, and Transformation with the Matching Pursuit Algorithm The matching pursuit (or MP) algorithm decomposes audio data into a collection of thousands of constituent sound particles or gaborets. These particles correspond to the “quantum” or granular model of sound posited by Dennis Gabor. This robust and highresolution analysis technique creates new possibilities for sound visualization and transformation. This paper presents an account of a first round of experiments with MP-based visualization and transformation techniques.
Download Removing Crackle from an LP Record via Wavelet Analysis The familiar “crackling” is one of the undesirable phenomena which we deal with in an LP record. Wavelet analysis brings a new alternative approach to the removal of this feature in the restoration process of the recording. In the paper, the principle of this method is described. A theoretical discussion of how the selection of the wavelet basis affects the quality of the restoration is also included.
Download Spectral Delays with Frequency Domain Processing In this paper the author presents preliminary research undertaken on spectral delays using frequency domain processing. A Max/MSP patch is presented in which it is possible to delay individual bins of a Fourier transform and several musically interesting applications of the patch, including the ability to create distinct spatial images and spectral trajectories are outlined.
Download Exponential Weighting Method for Sample-by-Sample Update of Warped AR-Model Auto-regressive (AR) modeling is a powerful tool having many ap plications in audio signal processing. The modeling procedure can be focused to low or high frequency range using frequency warp ing. Conventionally the AR-modeling procedure is accomplished with frame-by-frame processing which introduces latency. As with any frame-by-frame algorithm full frame has to be available for the algorithm before any output can be produced. This latency makes AR-modeling more or less unusable in real-time sound effects es pecially when long frame lengths are required. In this paper we introduce an exponential weighting (EW) method for sample-bysample update of the warped AR-model. This method reduces the latency down to the order of the AR-model.
Download Emulating Rough and Growl Voice in Spectral Domain This paper presents a new approach on transforming a modal voice into a rough or growl voice. The goal of such transformations is to be able to enhance voice expressiveness in singing voice productions. Both techniques work with spectral models and are based on adding sub-harmonics in frequency domain to the original input voice spectrum.
Download AIDE, A New Digital Audio Effects Development Environment This paper describes a new rapid development environment for digital audio applications and computer instruments, AIDE (Audio Instrument Development Environment). The system is designed to help users build signal processing applications for use in music, multimedia and sound design. Based on a graphical patching principle, this system generates software using the V and Sound Object libraries. These provide the graphical interface/application framework and sound processing elements, respectively, for standalone programmes generated by AIDE. It is envisaged that the system will also generate application components in addition to stand-alone programs. The paper outlines in some detail the elements involved in the software. It discusses how the system is aimed at different types of users with different levels of interaction. The paper concludes with an overview of the typical application development cycle using the system.
Download ADAM - A 64 Channel General Purpose Real-Time Audio Signal Processor In this paper we introduce a 64 channel audio processing unit made in our department. The audio processor uses a 16 bit control unit (Infineon XC 167) with ethernet interface running a realtime operating system and two Analog Devices ADSP-TS101S high performance tigerSHARC DSPs for audio stream processing. The first project implemented on this equipment is a 64 channel in, 32 channel out audio mixer with a sampling frequency of 48 kHz (leaving another 32 channels for effect feedback loops) or 96 kHz, alternatively. The audio processor is fully remote controllable via TCP/IP.
Download Non-Linear Digital Implementation of the Moog Ladder Filter This paper presents a non-linear digital implementation of the Moog ladder filter. The implementation is relatively efficient and suitable for inclusion into real-time systems, for example virtual analog synthesizers. The analog circuit is analyzed to produce a differential equation. This equation is solved using Euler’s method, and the result is shown to be equivalent to a cascade of first order IIR sections with embedded non-linearities. Finally, the filter structure is modified to improve tuning.