AES London 2010
Poster Session P10

Sunday, May 23, 09:00 — 10:30 (Room C4-Foyer)

P10 - Audio Processing—Analysis and Synthesis of Sound

P10-1 Cellular Automata Sound Synthesis with an Extended Version of the Multitype Voter Model—Jaime Serquera, Eduardo R. Miranda, University of Plymouth - Plymouth, UK
In this paper we report on the synthesis of sounds with cellular automata (CA), specifically with an extended version of the multitype voter model (MVM). Our mapping process is based on DSP analysis of automata evolutions and consists in mapping histograms onto sound spectrograms. This mapping allows a flexible sound design process, but due to the non-deterministic nature of the MVM such process acquires its maximum potential after the CA run is finished. Our extended version model presents a high degree of predictability and controllability making the system suitable for an in-advance sound design process with all the advantages that this entails, such as real-time possibilities and performance applications. This research focuses on the synthesis of damped sounds.
Convention Paper 8029 (Purchase now)

P10-2 Stereophonic Rendering of Source Distance Using DWM-FDN Artificial Reverberators—Saul Maté-Cid, Hüseyin Hacihabiboglu, Zoran Cvetkovic, King's College London - London, UK
Artificial reverberators are used in audio recording and production to enhance the perception of spaciousness. It is well known that reverberation is a key factor in the perception of the distance of a sound source. The ratio of direct and reverberant energies is one of the most important distance cues. A stereophonic artificial reverberator is proposed that allows panning the perceived distance of a sound source. The proposed reverberator is based on feedback delay network (FDN) reverberators and uses a perceptual model of direct-to-reverberant (D/R) energy ratio to pan the source distance. The equivalence of FDNs and digital waveguide mesh (DWM) scattering matrices is exploited in order to devise a reverberator relevant in the room acoustics context.
Convention Paper 8030 (Purchase now)

P10-3 Separation of Music+Effects Sound Track from Several International Versions of the Same Movie—Antoine Liutkus, Télécom ParisTech - Paris, France; Pierre Leveau, Audionamix - Paris, France
This paper concerns the separation of the music+effects (ME) track from a movie soundtrack, given the observation of several international versions of the same movie. The approach chosen is strongly inspired from existing stereo audio source separation and especially from spatial filtering algorithms such as DUET that can extract a constant panned source from a mixture very efficiently. The problem is indeed similar for we aim here at separating the ME track, which is the common background of all international versions of the movie soundtrack. The algorithm has been adapted to a number of channels greater than 2. Preprocessing techniques have also been proposed to adapt the algorithm to realistic cases. The performances of the algorithm have been evaluated on realistic and synthetic cases.
Convention Paper 8031 (Purchase now)

P10-4 A Differential Approach for the Implementation of Superdirective Loudspeaker Array—Jung-Woo Choi, Youngtae Kim, Sangchul Ko, Jungho Kim, SAIT, Samsung Electronics Co. Ltd. - Gyeonggi-do, Korea
A loudspeaker arrangement and corresponding analysis method to obtain a robust superdirective beam are proposed. The superdirectivity technique requires precise matching of the sound sources modeled to calculate excitation patterns and those used for the loudspeaker array. To resolve the robustness issue arising from the modeling mismatch error, we show that the overall sensitivity to the model-mismatch error can be reduced by rearranging loudspeaker positions. Specifically, a beam pattern obtained by a conventional optimization technique is represented as a product of robust delay-and-sum patterns and error-sensitive differential patterns. The excitation pattern driving the loudspeaker array is then reformulated such that the error-sensitive pattern is only applied to the outermost loudspeaker elements, and the array design that fits to the new excitation pattern is discussed.
Convention Paper 8032 (Purchase now)

P10-5 Improving the Performance of Pitch Estimators—Stephen J. Welburn, Mark D. Plumbley, Queen Mary University of London - London, UK
We are looking to use pitch estimators to provide an accurate high-resolution pitch track for resynthesis of musical audio. We found that current evaluation measures such as gross error rate (GER) are not suitable for algorithm selection. In this paper we examine the issues relating to evaluating pitch estimators and use these insights to improve performance of existing algorithms such as the well-known YIN pitch estimation algorithm.
Convention Paper 8033 (Purchase now)

P10-6 Reverberation Analysis via Response and Signal Statistics—Eleftheria Georganti, Thomas Zarouchas, John Mourjopoulos, University of Patras - Patras, Greece
This paper examines statistical quantities (i.e., kurtosis, skewness) of room transfer functions and audio signals (anechoic, reverberant, speech, music). Measurements are taken under various reverberation conditions in different real enclosures ranging from small office to a large auditorium and for varying source–receiver positions. Here, the statistical properties of the room responses and signals are examined in the frequency domain. From these properties, the relationship between the spectral statistics of the room transfer function and the corresponding reverberant signal are derived.
Convention Paper 8034 (Purchase now)

P10-7 An Investigation of Low-Level Signal Descriptors Characterizing the Noise-Like Nature of an Audio Signal—Christian Uhle, Fraunhofer Institute for Integrated Circuits IIS - Erlangen, Germany
This paper presents an overview and an evaluation of low-level features characterizing the noise-like or tone-like nature of an audio signal. Such features are widely used for content classification, segmentation, identification, coding of audio signals, blind source separation, speech enhancement, and voice activity detection. Besides the very prominent Spectral Flatness Measure various alternative descriptors exist. These features are reviewed and the requirements for these features are discussed. The features in scope are evaluated using synthetic signals and exemplarily real-world application related to audio content classification, namely voiced-unvoiced discrimination for speech signals and speech detection.
Convention Paper 8035 (Purchase now)

P10-8 Algorithms for Digital Subharmonic Distortion—Zlatko Baracskai, Ryan Stables, Birmingham City University - Birmingham, UK
This paper presents a comparison between existing digital subharmonic generators and a new algorithm developed with the intention of having a more pronounced subharmonic frequency and reduced harmonic, intermodulation and aliasing distortions. The paper demonstrates that by introducing inversions of a waveform at the minima and maxima instead of the zero crossings, the discontinuities are mitigated and various types of distortion are significantly attenuated.
Convention Paper 8036 (Purchase now)