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7th International Conference on Digital audio Effects, DAFx 2004

作者： Merimaa, Juha Pulkki, Ville Institute of Communication Acoustics Ruhr-Universität Bochum Germany Lab. of Acoustics and Audio Signal Processing Helsinki University of Technology Finland

Spatial Impulse Response Rendering (SIRR) is a recent technique for reproduction of room acoustics with a multichannel loudspeaker system. SIRR analyzes the direction of arrival and diffuseness of measured room responses within frequency bands. Based the analysis data, a multichannel response suitable for reproduction with any chosen surround loudspeaker setup is synthesized. When loaded to a convolving reverberator, the synthesized responses create a very natural perception of space corresponding to the measured room. In this paper, the SIRR method is described and listening test results are reviewed. The sound intensity based analysis is refined, and improvements for the synthesis of diffuse time-frequency components are discussed. © 2004 Proceedings of the International Conference on Digital audio Effects, DAFx. All rights reserved.

关键词： Loudspeakers

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High quality harmonic excitation linear predictive speech coding at 2 kb/s

High quality harmonic excitation linear predictive speech co...

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International Symposium on Chinese Spoken Language processing

作者： Changchun Bao J. Lukaside C. Ritz Speech and Audio Signal Processing Lab Beijing University of Technology Beijing China Whisper Lab TITR University of Wollongong NSW Australia

ISBN: (纸本)0780386787

The paper presents a high quality harmonic excitation linear predictive (HE-LPC) speech coder operating at 2 kb/s based on a harmonic excitation model with two bands. The system incorporates novel features such as: combined pitch detection; residual harmonic matching voicing determination; extraction and interpolation of residual harmonic magnitudes. Subjective listening tests indicate that this coder has the same quality as that of the Federal Standard MELP (mixed excitation linear prediction) coder at 2.4 kb/s, whether the training database is from Chinese or English.

关键词： Speech coding Delay estimation Speech analysis Speech processing Interpolation Linear predictive coding Frequency domain analysis Vocoders Quantization Performance analysis

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Linear Predictive method with Low-Frequency Emphasis 8

Linear Predictive method with Low-Frequency Emphasis

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8th European Conference on Speech Communication and Technology, EUROSPEECH 2003

作者： Alku, Paavo Bäckström, Tom Lab. of Acoustics and Audio Signal Processing Helsinki University of Technology Finland

An all-pole modeling technique, Linear Prediction with Low-frequency Emphasis (LPLE), which emphasizes the lower frequency range of speech, is presented. The method is based on first interpreting conventional linear predictive (LP) analyses of successive prediction orders with parallel structures using the concept of symmetric linear prediction. In these implementations, symmetric linear prediction is preceded by simple pre-filters, which are of either low or high frequency characteristics. Combining those symmetric linear predictors that are not preceded by high-frequency pre-filters yields the LPLE predictor. It is proved that the all-pole filters computed by LPLE are always stable. The results show that the method is well-suited when low-order all-pole models with improved modeling of the lowest formants are needed.

关键词： Forecasting

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INTERPOLATION OF LONG GAPS IN audio signalS USING THE WARPED BURG'S METHOD 6

INTERPOLATION OF LONG GAPS IN AUDIO SIGNALS USING THE WARPED...

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6th International Conference on Digital audio Effects, DAFx 2003

作者： Esquef, Paulo A.A. Välimäki, Vesa Roth, Kari Kauppinen, Ismo Lab. of Acoustics and Audio Signal Processing Helsinki University of Technology Finland Lab. of Optics and Spectroscopy University of Turku Department of Physics Finland

This paper addresses the reconstruction of missing samples in audio signals via model-based interpolation schemes. We demonstrate through examples that employing a frequency-warped version of Burg's method is advantageous for interpolation of long duration signal gaps. Our experiments show that using frequency-warping to focus modeling on low frequencies allows reducing the order of the autoregressive models without degrading the quality of the reconstructed signal. Thus a better balance between qualitative performance and computational complexity can be achieved. © *** rights reserved.

关键词： Interpolation

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Detection of clicks in audio signals using warped linear prediction 14

Detection of clicks in audio signals using warped linear pre...

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14th International Conference on Digital signal processing, DSP 2002

作者： Esquef, Paulo Karjalainen, Matti Välimäki, Vesa Lab. of Acoustics and Audio Signal Processing Helsinki University of Technology P.O.Box 3000 HUT EspooFIN-02015 Finland Pori School of Technology and Economics Tampere University of Technology P.O.Box 300 PoriFIN-28101 Finland

ISBN: (纸本)0780375033

In this work warped linear prediction (WLP) is applied to a model-based method to detect impulsive disturbances in audio signals. According to simulations performed on artificially corrupted audio signals the adoption of negative values for the warping factor favors the click detection scheme. As a consequence, for equal levels of missing (false) detection the WLP-based scheme yields consistently lower percentage of false (missing) detection than the conventional method. © 2002 IEEE.

关键词： signal detection

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One-delayed-mass model for efficient synthesis of glottal flow 7

One-delayed-mass model for efficient synthesis of glottal fl...

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7th European Conference on Speech Communication and Technology - Scandinavia, EUROSPEECH 2001

作者： Avanzini, Federico Alku, Paavo Karjalainen, Matti Dipartimento di Elettronica Ed Informatica Università di Padova Italy Lab. of Acoustics and Audio Signal Processing Helsinki University of Technology Finland

ISBN: (纸本)8790834100

A lumped physical model of the glottal source is presented. Vocal folds are described as single masses, but vertical phase differences between upper and lower margins of the folds are taken into account by appropriately describing the non-linear interaction of the mechanical model with aerodynamics. This results in a modified one-mass model, or a "one-delayed-mass model". Analysis on numerical simulations shows that the system behaves qualitatively as higher-dimensional models (such as the two-mass model by Ishizaka and Flanagan);in particular, control over flow skewness is guaranteed, allowing for synthesis of realistic glottal flow waveforms. As only one degree of freedom (one mass) is needed in the model, structure and number of parameters are drastically reduced, thus making it suitable for real-Time synthesis applications.

关键词： Degrees of freedom (mechanics)

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Tone hole radiation directivity measurements

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The Journal of the Acoustical Society of America 2001年第5_SUPPLEMENT期110卷 2754-2754页

作者： Gary P. Scavone Matti Karjalainen CCRMA Dept. of Music Stanford Univ. Stanford CA 94305 Lab. of Acoust. and Audio Signal Processing Helsinki Univ. of Technology FIN‐02015 HUT Finland

Measurements have been conducted in an anechoic chamber for a comparison to current acoustic theory with regard to radiation directivity from a cylindrical pipe with toneholes. Several difficulties arise in measurements of this sort, including (1) the generation of sufficient driving signal strength at the pipe input for pickup by an external microphone; (2) external source‐to‐pickup isolation; (3) measurement contamination due to nonlinear driver distortion. Time‐delay spectrometry using an exponentially swept sine signal was employed to determine impulse responses at points external to the experimental air column. This technique is effective in clearly isolating nonlinear artifacts from the desired linear system response along the time axis, thus allowing the use of a strong driving signal without fear of nonlinear distortion. The experimental air column was positioned through a wall conduit into the anechoic chamber such that the driver and pipe input were located outside the chamber while the open pipe end and tone holes were inside the chamber, effectively isolating the source from the pickup. Measured results are compared to both transmission‐line, frequency–domain simulations, as well as time–domain digital waveguide calculations.

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INTERPOLATED AND WARPED 2-D DIGITAL WAVEGUIDE MESH ALGORITHMS

INTERPOLATED AND WARPED 2-D DIGITAL WAVEGUIDE MESH ALGORITHM...

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2000 COST G-6 Conference on Digital audio Effects, DAFx 2000

作者： Välimäki, Vesa Savioja, Lauri Lab. of Acoustics and Audio Signal Processing Helsinki University of Technology Finland Telecommun. Software and Multimedia Lab Helsinki University of Technology Finland

Interpolated and warped digital waveguide mesh algorithms have been developed to overcome the problem caused by direction and frequency-dependence of wave travel speed in digital waveguide mesh simulations. This paper reviews the interpolation methods applicable in the two-dimensional case. The bilinear interpolation technique and two other approaches are briefly recapitulated. The use of 2-D quadratic interpolation results in a mesh structure that is efficiently implemented using only additions and binary shifts. The optimized interpolated 2-D mesh is obtained by choosing the free parameters of the interpolation method so that the difference in wave travel speed in the axial and diagonal directions is minimized. The output signal of any digital waveguide mesh can be processed with a warped FIR filter to reduce the dispersion error at all frequencies. The frequency-warping factor must be optimized individually for each mesh structure. We introduce an extended method called multiwarping, which can be used to further decrease the dispersion error. The frequency range of digital waveguide mesh simulations is also discussed. © Proceedings of the International Conference on Digital audio Effects, DAFx 2000.

关键词： FIR filters

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METHODS FOR MODELING REALISTIC PLAYING IN PLUCKED-STRING SYNTHESIS: ANALYSIS, CONTROL AND SYNTHESIS

METHODS FOR MODELING REALISTIC PLAYING IN PLUCKED-STRING SYN...

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2000 COST G-6 Conference on Digital audio Effects, DAFx 2000

作者： Laurson, Mikael Erkut, Cumhur Välimäki, Vesa Center for Music and Technology Sibelius Academy Helsinki Finland Lab. of Acoustics and Audio Signal Processing Helsinki Univ. of Technology Espoo Finland

This paper summarizes our recent achievements in developing signal processing methods for model-based synthesis of plucked-string instruments. The methods discussed are applicable to a broad family of musical instruments, but this paper concentrates on a case study where we focus on the classical guitar. The real-time guitar synthesis engine is based on digital waveguide modeling and commuted waveguide synthesis. The paper provides an overview of our Lisp-based real-time synthesizer and how it is controlled. Furthermore, extended procedures to synthesize transients, pizzicati, and improved dynamics are presented. © Proceedings of the International Conference on Digital audio Effects, DAFx 2000.

关键词： Musical instruments

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Virtual acoustics - Applications and technology trends 10

Virtual acoustics - Applications and technology trends

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2000 10th European signal processing Conference, EUSIPCO 2000

作者： Huopaniemi, Jyri Savioja, Lauri Lokki, Tapio Väänänen, Riitta Nokia Research Center Speech and Audio Systems Lab Finland Helsinki Univ. of Tech. Telecommunications Software and Multimedia Laboratory Finland Helsinki Univ. of Tech. Laboratory of Acoustics and Audio Signal Processing Finland

ISBN: (纸本)9789521504433

Virtual acoustics is a general term for the modeling of acoustical phenomena and systems with the aid of a computer. It comprises many different fields of acoustics, but in the context of this paper the term is restricted to describe a system ranging from sound source and acoustics modeling in rooms to spatial auditory perception simulation in humans. In other words, virtual acoustics concept covers three major subsystems in acoustical communication: source, transmission, and the listener. This paper discusses virtual acoustics from the point of view of applications and technology trends. © 2000 EUSIPCO.

关键词： signal processing

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