Saturday, October 21, 2:00 pm — 4:30 pm (Rm 1E11)
Chair:
Joerg Panzer, R&D Team - Salgen, Germany
P18-1 Dynamic Diffuse Signal Processing for Low-Frequency Spatial Variance Minimization across Wide Audience Areas—Jonathan Moore, University of Derby - Derby, UK; Adam J. Hill, University of Derby - Derby, Derbyshire, UK; Gand Concert Sound - Elk Grove Village, IL, USA
Diffuse signal processing (DiSP) is a method of decorrelating coherent audio signals that is applicable to various components of sound reinforcement systems. Previous tests have indicated that DiSP can successfully decorrelate multiple low-frequency sources, leading to the reduction of comb filtering effects. However, results also show that performance is variable with source material and that effectiveness is reduced in closed acoustic spaces. In this work a dynamic variant of DiSP is examined where the decorrelation algorithm varies over time. The effectiveness of the processing is analyzed and compared to static DiSP and unprocessed systems. Results show that dynamic DiSP provides superior low-frequency spatial variance reduction to static DiSP due to improved decorrelation between direct sounds and early reflections.
P18-2 A Novel Procedure for Direct-Method Measurement of the Full-Matrix Speech Transmission Index—Jan A. Verhave, Embedded acoustics BV - Delft, The Netherlands; Sander van Wijngaarden, Embedded acoustics BV - Delft, The Netherlands
When measuring the Speech Transmission Index (STI), until now one had to choose between two alternatives: impulse-response based full STI measurements (indirect method), or measurements based on modulated STIPA signals (direct method). Limitations apply when using either method. A novel procedure is proposed to measure the full STI through the direct method. The procedure combines advantages of indirect full STI measurements and direct STIPA measurements, completing a full STI measurement in 65.52 seconds. Similar to STIPA, the test signal is simultaneously modulated with 2 modulation frequencies per octave band. However, a rotation scheme is applied that uses a different set of modulation frequencies during different stages of the measurement, ending up with a full matrix (7 octaves x 14 modulation frequencies).
P18-3 Blind Estimation of the Reverberation Fingerprint of Unknown Acoustic Environments—Prateek Murgai, Center for Computer Research in Music and Acoustics (CCRMA), Stanford University - Palo Alto, CA, USA; Mark Rau, Center for Computer Research in Music and Acoustics (CCRMA), Stanford University - Palo Alto, CA, USA; Jean-Marc Jot, Magic Leap - Sunnyvale, CA, USA
Methods for blind estimation of a room’s reverberation properties have been proposed for applications including speech dereverberation and audio forensics. In this paper, we study and evaluate algorithms for online estimation of a room’s “reverberation fingerprint”, defined by its volume and its frequency-dependent diffuse reverberation decay time. Both quantities are derived adaptively by analyzing a single-microphone reverberant signal recording, without access to acoustic source reference signals. The accuracy and convergence of the proposed techniques is evaluated experimentally against the ground truth obtained from geometric and impulse response measurements. The motivations of the present study include the development of improved headphone 3D audio rendering techniques for mobile computing devices.
P18-4 Microphone Selection Based on Direct to Reverberant Ratio Estimation—Alexis Favrot, Illusonic GmbH - Zürich, Switzerland; Christof Faller, Illusonic GmbH - Uster, Zürich, Switzerland; EPFL - Lausanne, Switzerland
Microphone recording in a room is ideally carried out by using a close distance microphone to prevent reverberation and noise annoyances, but this restricts the flexibility and the surface covered by the recording. When using multiple distant microphones, a microphone selection algorithm is needed for selecting the momentarily best microphone, namely the one with the least reverberation. Given several microphones arbitrarily distributed in a room, this paper describes an algorithm which, based on an estimation of the direct-to-reverberation ratio for each microphone, switches to the best microphone. The algorithm allows prioritizing a microphone and compensation of different directivity patterns.
P18-5 Experimental Investigation on Varied Degrees of Sound Field Diffuseness in Full Scale Rooms—Alejandro Bidondo, Universidad Nacional de Tres de Febrero - UNTREF - Caseros, Buenos Aires, Argentina; Sergio Vazquez, Universidad Nacional de Tres de Febrero - Buenos Aires, Argentina; Javier Vazquez, Universidad Nacional de Tres de Febrero - Buenos Aires, Argentina
Sound field diffusion in enclosures should be experimentally quantified based on measured room impulse responses, at least to know how many scattering surfaces produce a sufficiently diffuse sound field for each application. To achieve this a parameter, the Sound Field Diffusion Coefficient (SFDC) which is still under development, was applied. SFDC expresses the reflection's amplitude control and temporal distribution gaussianity, using third octave-band energy-decay compensated impulse responses and taking reference with SFDC average results from a set of impulse responses synthesized with Gaussian white noise. In an attempt to demonstrate the quantification capability of the SFDC, a systematic investigation was conducted whereby varied room configurations using carefully designed scattered interior surfaces were examined with the hypothesis that varied degrees of surface scattering will ultimately lead to varied degrees of sound field diffusion inside two, full scale, rooms. To this end, each room´s floor was covered with different configurations ranging from no diffusers to 16.74 m2 of diffusely reflecting surfaces, in 3 steps. This paper discusses the experimental design and evaluates the results of data collected using systematic modifications of varied degrees of surface scattering, each with combinations of different source orientations and microphone positions.