Download An Eyes-free User Interface Controlled by Finger Snaps
A novel way of controlling a simple user interface based on detecting and localizing finger snaps of the user is presented. The analysis method uses binaural signals recorded from the ears of the user. Transient sounds are first detected from a continuous audio stream, followed by cross-correlation based localization and simple band-energy ratio based classification. The azimuth plane around the user is divided into three sectors, each of which corresponds to one of the three “buttons” in the interface. As an example, the interface is applied for controlling the playlist of an MP3 player. The algorithm performance was evaluated using a real-world recording. While the algorithm looks promising, more research is needed before it is ready for commercial applications.
Download Granular tools for real-time sound processing, Examples of mapping with video
Download Analysis-assisted sound processing with audiosculp
Digital audio effects using phase vocoder techniques are currently in widespread use. However, their interfaces often hide vital parameters from the user. This fact, and the generally limited ways in which sound designing and compositional tools can represent sounds and their spectral content, complicates the effective use of the full potential of modern effect processing algorithms. This article talks about ways in which to use analysis to obtain better processing results with phase vocoder based effects, and how these techniques are implemented in IRCAM's AudioSculpt application. Also discussed are the advantages of using the SDIF format to exchange analysis data between various software components, which facilitates the integration of new analysis and processing algorithms.
This paper intends to describe the desirable features of a complete, powerful and highly customizable real-time audio algorithm implementation system, and to provide the guidelines to its implementation. The goal is to design a platform by means of which new sophisticated audio algorithms can be developed, tested and used with the minimum effort. The idea is to build large complex processing systems based on elemental building blocks which may interact in any possible manner. This way, by connecting existing, proved modules, such as filters, noise gates, or any new specific module, complex processes can be achieved and tested in a realtime environment with the minimum possible effort. The buildingblock philosophy would also make such a system very suitable for educational purposes, as it would make possible to ‘hear’ in realtime a particular complex algorithm with and without one of its blocks (a filter, for example), thus showing its importance.
Download Scalable Perceptual Mixing and Filtering of Audio Signals Using an Augmented Spectral Representation
Many interactive applications, such as video games, require processing a large number of sound signals in real-time. This paper proposes a novel perceptually-based and scalable approach for efficiently filtering and mixing a large number of audio signals. Key to its efficiency is a pre-computed Fourier frequency-domain representation augmented with additional descriptors. The descriptors can be used during the real-time processing to estimate which signals are not going to contribute to the final mixture. Besides, we also propose an importance sampling strategy allowing to tune the processing load relative to the quality of the output. We demonstrate our approach for a variety of applications including equalization and mixing, reverberation processing and spatialization. It can also be used to optimize audio data streaming or decompression. By reducing the number of operations and limiting bus traffic, our approach yields a 3 to 15-fold improvement in overall processing rate compared to brute-force techniques, with minimal degradation of the output.
Download An auditory 3D file manager designed from interaction patterns
This paper shows the design, implementation and evaluation of an auditory user interface for a file-manager application. The intention for building this prototype was to prove concepts developed to support user interface designers with design patterns in order to create robust and efficient auditory displays. The paper describes the motivation for introducing a mode-independent meta domain in which the design patterns were defined to overcome the problem of translating mainly visual concepts to the auditory domain. The prototype was implemented using the IEM Ambisonics libraries for Pure Data to produce high quality binaural audio rendering and used headtracking and a joystick as the main interaction devices.
Download Conjugate gradient techniques for multichannel acoustic echo cancellation
Conjugate Gradient (CG) techniques are suitable for resolution of time-variant system identification problems: adaptive equalization, echo cancellation, active noise cancellation, linear prediction, etc. These systems can be seen as optimization problems and CG techniques can be used to solve them. It has been demonstrated that, in the single-channel case, the conjugate gradient techniques provide a similar solution in terms of convergence rate than those provided by the recursive least square (RLS) method, involving higher complexity than the least mean square (LMS) but lower than RLS without stability issues. The advantages of these techniques are especially valuable in the case of high complexity and magnitude problems like multi-channel systems. This work develops CG algorithm for the adaptive MIMO (multiple-input and multiple-output) systems and tests it by solving a multichannel acoustic echo cancellation (MAEC) problem.
Download POWERWAVE: a high performance Single Chip Interpolating Wavetable Synthesizer
In this paper we introduce the single chip implementation of a 16 voices wavetable synthesizer. All digital functions (control and waveform generation) are contained in a single platform FPGA chip (Xilinx Virtex 2 Pro). Only the digital to analog conversion is done by a standard 96 kHz audio DAC (AD 1785). In the first version the synthesizer is controlled via standard RS 232 interface.
Download Estimating the amplitude of the cubic difference tone using a third order adaptive Volterra Filter
Design method of a nonlinear filter to estimate the amplitudes of cubic difference tones is presented. To this end, a third-order Volterra filter is used to model the nonlinearity of our auditory system, and the filter coefficients are obtained using an adaptive process. The results show the filtered outputs follow very closely the experimental data as the intensity levels and the frequencies of inputs vary especially when the frequency separation between the two primary tones is not large.
In this paper, we first define the scenario of a generic acoustic human/machine interface and then formulate the according fundamental signal processing problems. For signal reproduction, the requirements for ideal solutions are stated and some examples for the state of the technology are briefly reviewed. For signal acquisition, the fundamental problems ask for acoustic echo cancellation, desired source extraction, and source localization. After illustrating to which extent acoustic echo cancellation is already a solved problem, we present recent results for separation, dereverberation and localization of multiple source signals. As an underlying motivation for this synoptic treatment, we demonstrate that the considered subproblems (except localization) can be directly interpreted as signal separation or system identification problems with varying degrees of difficulty, which in turn determines the effectiveness of the known solutions.