Download Separation Of Speech Signal From Complex Auditory Scenes The hearing system, even in front of complex auditory scenes and in unfavourable conditions, is able to separate and recognize auditory events accurately. A great deal of effort has gone into the understanding of how, after having captured the acoustical data, the human auditory system processes them. The aim of this work is the digital implementation of the decomposition of a complex sound in separate parts as it would appear to a listener. This operation is called signal separation. In this work, the separation of speech signal from complex auditory scenes has been studied and an experimentation of the techniques that address this problem has been done.
Download EPS Models Of Am-Fm Vocoder Output For New Sounds Generations The Phase Vocoder [1] was originally introduced as an alternative approach to speech coding but has won much greater acceptance among the music community as a tool both for sound analysis and composition [2]. Although dormant for some time, there has been a resurgence of interest in AM-FM speech signal descriptions in the last ten years [3], [4]. With the intention of building on some of the new ideas proffered, the aim of this work is to first consider their application to musical signals. It then demonstrates how paramaterisation of the extracted AM-FM information using EPS (Exponential Polynomial Signal) models allows modification of the individual large- and small-grain features of the AM and FM components, thus providing a new way for generating audio effects.
Download Magnitude-Complementary Filters For Dynamic Equalization Discrete-time structures of first-order and second-order equalization filters are proposed. They turn to be particularly useful in applications where the equalization parameters are dynamically varied, such as in contexts of audio virtual reality. In fact, their design allows a simplified and more direct control of the filter coefficients, at the cost of some more computation cycles that are required, during each time step, by implementations on real-time processing devices.
Download TAPIIR, A Software Multitap The use of delays is one of the oldest techniques for effects processing and electro-acoustic composition[1]. Originally, tape-loops were used to create effects of echo and reverb. Nowadays, most hardware effect processors provide digital implementations. These have a clearly superior soundquality compared to tape-delays, but also imply some restrictions. Delay-length is limited by the internal memory, and delay time accuracy is often sacrificed for computational efficiency or even deliberately restricted for user-interfacing. This paper presents a dedicated software implementation of a flexible multi-delay, that aims to combine flexibility and high accuracy with high quality audio and usability. At the same time, several important issues in software effect processing will be addressed.
Download Time-Varying Filter In Non-Uniform Block Convolution This paper will describe further research on a real-time convolution algorithm for long a FIR filter based on nonuniform bock partitioning. The static behaviour of the algorithm which solves the dilemma between the computational load and the latency of the processing operation is well examined in literature. New directions are investigated to exploit the inherent features of the algorithm and utilise them for audio applications. Especially a dynamic exchange of filter coefficients or subsets of them of a room impulse response is discussed and implemented. Unlike to traditional DSP solutions the prototype is realised in portable software objects and components that can be compiled on multi-propose processing units like off-the-shelf computers with standard audio facilities and different operating systems. Keywords : convolution, spatial sound processing, real-time, room acoustics, sonification
Download Sound Morphing With Gaussian Mixture Models In this work a sound transformation model based on Gaussian Mixture Models is introduced and evaluated for audio morphing. To this aim, the GMM is used to build the acoustic model of the source sound, and a set of conversion functions, which rely on the acoustic model, is used to transform the source sound. The method is experimented on a set of monophonic sounds and results show that it provides promising features.
Download Voice source analysis for pitch-scale modification of speech signals Much research has shown that the voice source has strong influence on the quality of speech processing [4][5][6]. But in most of the existing speech modification algorithms, the effect of the voice source variation is neglected. This work explains why the existing modification scheme can’t truly reflect the voice source variation during pitch modification. We use synthesized voiced speech sound to compare an existing pitch modification scheme with our proposed voice source scaling based modification scheme. Results show that voice source scaling based pitch modification can be used for wider range pitch modification. Key word: speech pitch modification, voice source, formant synthesis.
Download Multiresolution Sinusoidal/Stochastic Model For Voiced-Sounds The goal of this paper is to introduce a complete analysis/resynthesis method for the stationary part of voiced-sounds. The method is based on a new class of wavelets, the Harmonic-Band Wavelets (HBWT). Wavelets have been widely employed in signal processing [1, 2]. In the context of sound processing they provided very interesting results in their first harmonic version: the Pitch Synchronous Wavelets Transform (PSWT) [3]. We introduced the Harmonic-Band Wavelets in a previous edition of the DAFx [4]. The HBWT, with respect to the PSWT allows one to manipulate the analysis coefficients of each harmonic independently. Furthermore one is able to group the analysis coefficients according to a finer subdivision of the spectrum of each harmonic, due to the multiresolution analysis of the wavelets. This allows one to separate the deterministic components of voiced sounds, corresponding to the harmonic peaks, from the noisy/stochastic components. A first result was the development of a parametric representation of the HBWT analysis coefficients corresponding to the stochastic components [5, 7]. In this paper we present the results concerning a parametric representation of the HBWT analysis coefficients of the deterministic components. The method recalls the sinusoidal models, where one models time-varying amplitudes and time varying phases [8, 9]. This method provides a new interesting technique for sound synthesis and sound processing, integrating a parametric representation of both the deterministic and the stochastic components of sounds. At the same time it can be seen as a tool for a parametric representation of sound and data compression.
Download Influence Of Frequency Distribution On Intensity Fluctuations Of Noise This article introduces mathematical parameters that generalize the frequency distribution of the statistical and spectral model. We show experiments that determine the influence of these parameters on the intensity fluctuations of the synthesized noise, accordingly to the theory developed by psycho-acoustic works. The main goal of these experiments is to be able to analyse and synthesize bands of noise with different spectral densities.
Download Scalable Spectral Reflections In Conic Sections The object of this project is to present a novel digital audio effect based on a real-time, windowed block-based FFT and inverse FFT. The effect is achieved by mirroring the spectrum, producing a sound effect ranging from a purer rendition of the original, through a rougher one, to a sound unrecognisable from the original. A mirror taking the shape of a conic section is constructed between certain partials, and the modified spectrum is created by reflecting the original spectrum in this mirror. The user can select the type and continuously vary the amount of curvature, typically ‘roughening’ the input sound quite gratifyingly. We demonstrate the system with live real-time audio via microphone.