120th AES Convention - Paris, France - Dates: Saturday May 20 - Tuesday May 23, 2006 - Porte de Versailles

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AES Paris 2006

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Last Updated: 20060418, mei

P12 - Signal Processing and High Resolution Audio, Part 2

Sunday, May 21, 14:00 — 18:00

Chair: Jan Abildgaard Pedersen, Lyngdorf Audio - Denmark

P12-1 SigmaStudio. A User-Friendly, Intuitive and Expandable, Graphical Development Environment for Audio/DSP ApplicationsMiguel Chavez, Camille Huin, Analog Devices, Inc. - Wilmington, MA, USA
Graphical development environments have been used in the audio industry for a number of years. Those who have fewer limitations have persisted and found a well-established pool of users that is reluctant to modify their design patterns and adopt different embedded processors and design environments. This paper provides a small history of the evolution of integrated development environments (IDEs). It then describes and explains the software architecture decisions and design challenges that were used to develop SigmaStudio. It will also show the advantages that those decisions have meant for the SigmaDSP family of audio-centric embedded processors.

[Associated Poster Presentation in Session P17, Monday, May 22, at 09:00]

Presentation is scheduled to begin at 14:00
Convention Paper 6714 (Purchase now)

P12-2 Filter Update Techniques for Adaptive Virtual Acoustic ImagingPal Mannerheim, Philip Nelson, University of Southampton - Southampton, UK; Youngtae Kim, Samsung Advanced Institute of Technology (SAIT) - Gyeonggi-do, Korea
This paper deals with filter updates for adaptive virtual acoustic imaging systems using binaural technology and loudspeakers. The problem is to update the inverse filters without creating any audible changes for the listener. The problem can be overcome by using either a very fine mesh for the inverse filters or by using commutation techniques.

Presentation is scheduled to begin t 14:20
Convention Paper 6715 (Purchase now)

P12-3 Adaptive Filters in Wavelet Transform DomainVladan Bajic, Audio-Technica US - Stow, OH, USA
This paper presents performance comparison between two methods of implementing adaptive filtering algorithms for noise reduction, namely the Normalized time domain Least Mean Squares (NLMS) algorithm and the Wavelet transform domain LMS (WLMS). A brief theoretical development of both methods is explained, and then both algorithms are implemented on a real-time Digital Signal Processing (DSP) system used for audio signals processing. Results are presented showing the performance of each algorithm both in time and frequency domains. Noise reduction effects produced by different algorithms were shown across the spectrum, and distorting effects were analyzed. Trade-offs of convergence speed versus added noise were analyzed. Overall results show convergence speed improvement when using WLMS algorithms over the NLMS algorithm.

[Associated Poster Presentation in Session P17, Monday, May 22, at 09:00]

Presentation is scheduled to begin at 14:40
Convention Paper 6716 (Purchase now)

P12-4 Adaptive Time-Frequency Resolution for Analysis and Processing of AudioAlexey Lukin, Moscow State University - Moscow, Russia; Jeremy Todd, iZotope, Inc. - Cambridge, MA, USA
Filter banks with fixed time-frequency resolution, such as the Short-Time Fourier Transform (STFT), are a common tool for many audio analysis and processing applications allowing effective implementation via the Fast Fourier Transform (FFT). The fixed time-frequency resolution of the STFT can lead to the undesirable smearing of events in both time and frequency. In this paper we suggest adaptively varying STFT time-frequency resolution in order to reduce filter bank-specific artifacts while retaining adequate frequency resolution. Several strategies for systematic adaptation of time-frequency resolution are proposed. The introduced approach is demonstrated as applied to spectrogram displays, noise reduction, and spectral effects processing.

[Associated Poster Presentation in Session P17, Monday, May 22, at 09:00]

Presentation is scheduled to begin at 15:00
Convention Paper 6717 (Purchase now)

P12-5 Advanced Methods for Shaping Time-Frequency Areas for the Selective Mixing of SoundsPiotr Kleczkowski, AGH University of Science and Technology - Krakow, Poland; Adam Kleczkowski, University of Cambridge - Cambridge, UK
The “Selective Mixing of Sounds” (AES 119th Convention Paper 6552) contains a large and conceptually challenging part, which had not been developed previously. This is a method of determining the areas of dominance by different tracks in the time-frequency plane. It has a major effect on the overall quality of the sound. In this paper we propose and compare a range of appropriate algorithms. We begin with a simple two-dimensional running mean combined with a rule selecting the track characterized by the maximum energy, followed by a low-pass filter based on the two-dimensional Fourier transform. We also propose two novel methods based on the Monte-Carlo approach, in which local probabilistic rules are iterated many times to produce a required level of smoothing.

[Associated Poster Presentation in Session P17, Monday, May 22, at 09:00]

Presentation is scheduled to begin at 15:20
Convention Paper 6718 (Purchase now)

P12-6 Demixing Commercial Music Productions via Human-Assisted Time-Frequency MaskingMarc Vinyes, Jordi Bonada, Alex Loscos, Pompeu Fabra University - Barcelona, Spain
Audio blind separation in real commercial music recordings is still an open problem. In the last few years some techniques have provided interesting results. This paper presents a human-assisted clusterization of the DFT coefficients for the time-frequency masking demixing technique. The DFT coefficients are grouped by adjacent pan, interchannel phase difference, and magnitude and magnitude-variance with a real-time interactive graphical interface. Results prove that an implementation of such technique can be used to demix tracks from nowadays commercial songs. Sample sounds can be found at http://www.iua.upf.es/~mvinyes/abs/demos.

[Associated Poster Presentation in Session P17, Monday, May 22, at 09:00]

Presentation is scheduled to begin at 15:40
Convention Paper 6719 (Purchase now)

P12-7 A Multichannel Speech Dereverberation Technique Based Upon the Wiener FilterDenis McCarthy, Frank Boland, Trinity College - Dublin, Ireland
We present a new method for dereverberating speech based upon a multichannel Wiener Filter and a microphone array. We demonstrate the effectiveness of this method under real, reverberant conditions and show that the method may be described as a self-steering beamformer. Furthermore, we investigate the performance of the method under simulated conditions, designed to closely match the acoustic characteristics of the real room environment. These simulations yield significantly inferior results to those obtained using real recordings and we show that this is a result of the failure of simulated impulse responses to accurately model real impulse responses in certain critical respects.

Presentation is scheduled to begin at 16:00
Convention Paper 6720 (Purchase now)

P12-8 Effective Room Equalization Based on Warped Common Acoustical Poles and ZerosJunho Lee, Jae-woong Jeong, Yonsei University - Seoul, Korea; Young-cheol Park, Yonsei University - Wonju-City, Gangwon-Do, Korea; Seh-Woong Jeong, Samsung Electronics Co., Ltd. - Yongin-City, Gyeonggi-Do, Korea; Dae-hee Youn, Yonsei University - Seoul, Korea
This paper presents a new method of designing room equalization filters using a warped common acoustical pole and zero (WCAPZ) modeling. The proposed method is capable of significantly reducing the order of the equalization filters without sacrificing the filter performance, especially at low frequencies. Thus, the associated input-output delay is much smaller than the conventional block transform method while its computational complexity is comparable to it. The computational complexity also is still comparable to the conventional room equalization method, since the filter is implemented in the linear frequency domain after the pole-zero dewarping. Simulation results confirm that the use of the proposed algorithm significantly improves the room equalization over a range of low frequencies.

Presentation is scheduled to begin at 16:20
Convention Paper 6721 (Purchase now)

P12-9 Parametric Recursive Higher-Order Shelving FiltersMartin Holters, Udo Zölzer, Helmut-Schmidt-University - Hamburg, Germany
The main characteristic of shelving filters, as commonly used in audio equalization, is to amplify or attenuate a certain frequency band by a given gain. For parametric equalizers, a filter structure is desirable that allows independent adjustment of the width and center frequency of the band, and the gain. In this paper we present a design for arbitrary-order shelving filters and a suitable parametric structure. A low shelving filter design based on Butterworth filters is decomposed such that the gain can be easily adjusted. Transformation to the digital domain is performed, keeping gain and denormalized cut-off frequency independently controllable. Finally, we obtain mid- and high-shelving filters using a simple manipulation, providing the desired parametric filter structure.

Presentation is scheduled to begin t 16:40
Convention Paper 6722 (Purchase now)

P12-10 Enhanced Control of Sound Field Radiated by Co-Axial Loudspeaker Systems Using Digital Signal Processing TechniquesHmaied Shaiek, ENST de Bretagne - Brest Cedex, France; Bernard Debail, Cabasse Acoustic Center - Plouzané, France; Jean Marc Boucher, ENST de Bretagne - Brest Cedex, France; Yvon Kerneis, Pierre Yves Diquelou, Cabasse Acoustic Center - Plouzané, France
In multiway loudspeaker systems, digital signal processing techniques have been used so far mainly to correct frequency response, time alignment, and out of axis lobbing. In this paper a dedicated signal processing technique is described in order to also control the sound field radiated by co-axial loudspeaker systems in the overlap frequency band of drivers. Trade-offs and practical constraints (crossover, time shift, gain, etc.) are discussed and an optimization algorithm is proposed to provide the best achievable result. Real-time implementation of this technique is presented and leads to a nearly ideal point source.

[Associated Poster Presentation in Session P17, Monday, May 22, at 09:00]

Presentation is scheduled to begin at 17:00
Convention Paper 6723 (Purchase now)

P12-11 Network Music Performance (NMP) in Narrow Band NetworksAlexander Carôt, International School of New Media (ISNM) - Lübeck, Germany; Ulrich Krämer, Gerald Schuller, Fraunhofer Institute for Digital Media Technology - Ilmenau, Germany
Playing live music on the Internet is one of the hardest disciplines in terms of low delay audio capture and transmission, time synchronization, and bandwidth requirements. This has already been successfully evaluated with the Soundjack software, which can be described as a low latency UDP streaming application. In combination with the new Fraunhofer ULD Codec this technology could now be used in narrow band DSL networks without a significant increase of latency. This paper first describes the essential basics of network music performances in terms of soundcard and network issues and finally reviews the context under DSL narrow band network restrictions and the usage of the ULD Codec.

[Associated Poster Presentation in Session P17, Monday, May 22, at 09:00]

Presentation is scheduled to begin at 17:20
Convention Paper 6724 (Purchase now)

P12-12 Intensive Noise Reduction Utilizing Inharmonic Frequency Analysis of GHATeruo Muraoka, University of Tokyo - Komaba Meguro-ku, Tokyo, Japan; Ryuji Takamizawa, Matsushita Electric Industrial Co., Ltd. - Kadoma City, Osaka, Japan; Yoshihiro Kanda, Musashi Institute of Technology - Tamadutumi Setagaya, Tokyo, Japan; Takumi Ohta, Kenwood Corporation - Hachiouji City, Tokyo, Japan
Removal of noise in SP record reproduction were attempted utilizing GHA (Generalized Harmonic Analysis) as inharmonic frequency analysis. Spectrum subtraction is most common among conventional noise reduction techniques, however it has a side effect of musical noise generation. It is caused by inaccurate frequency resolution inherent to conventional harmonic frequency analysis. One method of inharmonic frequency analysis of GHA is equipped with excellent frequency resolution, and it has been put in practical use recently. The authors applied GHA for noise reduction and obtained better results than those by conventional spectrum subtraction. However, there still remained musical noise problems, and its major reason is spectral in-coincidence between pre-sampled reference noise and actually remained residual noise. The authors tried several countermeasures such as pre-spectral shaping of object signal and spectral similarity calculation of residual noise, etc. Through combining countermeasures, the authors achieved satisfactory noise reduction.

[Associated Poster Presentation in Session P17, Monday, May 22, at 09:00]

Presentation is scheduled to begin at 17:40
Convention Paper 6725 (Purchase now)

  (C) 2006, Audio Engineering Society, Inc.