Download Enhanced 3D sound field synthesis and reproduction system by compensating interfering reflections The antique stereophonic recording and playback format is going to be replaced by new surround sound formats in the near future. At the moment, various surround techniques are being investigated in many artistic and technical applications. The main concern is to find an appropriate recording and playback format which supports the natural spatial hearing cues. Therefore, surround sound systems should provide a homogeneous and coherent sound field image, both for recorded and synthesized sound fields [1]. In a homogeneous sound reproduction system, no direction is treated preferentially. Coherent sound field image means that the image remains stable under changes of the listener position, though the image may change as a natural sound field does. The Holophony and Ambisonic system described by Nicol and Emerit [2] is the basic approach. This system will be extended by a new approach to compensate the interfering reflections of the reproduction room. Further possibilities to determine higher order Ambisonic signals using the beam forming approach are investigated.
Download Multichannel sound reproduction system for binaural signals - the ambisonic approach Convincing sound reproduction via headphones requires filtering of virtual sound sources with head related transfer functions (HRTF). HRTFs describe signal differences at the two ear drums in level, time and frequency weighting [1,2]. Another psychoacoustic phenomenon of the human auditory system concerning hearing in natural sound fields is the improvement of source localization if small head movements are possible [3]. In this paper a computational efficient implementation to get dynamic binaural signals (in dependence on the head position) is proposed. This realtime modell is based on Ambisonic sound reproduction [4,5].
Download Further Investigations On 3D Sound Fields Using Distance Coding This investigation proposes a possibility to synthesise a true 3D sound field over loudspeakers. A new approach concerning the distance coding is presented. We tried to combine the benefits both using the Wave Field Synthesis (WFS) approach and Higher Order Ambisonics (HOA). Therefore the proposed system can be divided into two main parts. Firstly the determination of the driving functions of the sound sources using the WFS approach. Secondly the coding for transmission or storage whereby the scheme is based on the Ambisonics approach using higher orders. The paper is organised in three sections. The first section gives a brief introduction about the WFS and the HOA approaches. In the second section the derivation of the driving functions is presented and the coding scheme of the derived source signals is explained. Finally the paper is concluded and further possible research directions are identified.
Download Sound-System Design for a Professional Full-Flight Simulator In this paper, we present a sound system to be integrated in an accredited realistic full-flight simulator, used for the training of airline pilots. We discuss the design and implementation of a corresponding real-time signal-processing software providing threedimensional audio reproduction of the acoustic events on a flight deck. Here, the emphasis is on an aircraft of a specific type. We address issues of suitable data acquisition methods, and, most importantly, of functional signal analysis and synthesis techniques.
Download A New Functional Framework for a Sound System for Realtime Flight Simulation We will show a new sound framework and concept for realistic flight simulation. Dealing with a highly complex network of mechanical systems that act as physical sound sources the main focus is on a fully modular and extensible/scalable design. The prototype we developed is part of a fully functional Full Flight Simulator for Pilot Training.
Download Transaural Stereo in a Beamforming Approach This paper presents a study on algorithms for headphone-free binaural synthesis using a dedicated loudspeaker configuration. Both algorithms under investigation improve the properties of the binaural synthesis performance of the array. Firstly, beam-forming provides sound radiation localized at two freely adjustable, narrow target spots. Adjusting both spots to the locations of the listener’s ears achieves a good basis. Secondly, an additional interaural crosstalk canceler improves the overall result.
Download Chroma and MFCC Based Pattern Recognition in Audio Files Utilizing Hidden Markov Models And Dynamic Programming In this paper we present an algorithm to reveal the immanent musical structure within pieces of popular music. Our proposed model uses an estimate of the harmonic progression which is obtained by calculating beat-synchronous chroma vectors and letting a Hidden Markov Model (HMM) decide the most probable sequence of chords. In addition, MFCC vectors are computed to retrieve basic timbral information that can not be described by harmony. Subsequently, a dynamic programming algorithm is used to detect repetitive patterns in these feature sequences. Based on these patterns a second dynamic programming stage tries to find and link corresponding patterns to larger segments that reflect the musical structure.
Download Similarity-based Sound Source Localization with a Coincident Microphone Array This paper presents a robust, accurate sound source localization method using a compact, near-coincident microphone array. We derive features by combining the microphone signals and determine the direction of a single sound source by similarity matching. Therefore, the observed features are compared with a set of previously measured reference features, which are stored in a look-up table. By proper processing in the similarity domain, we are able to deal with signal pauses and low SNR without the need of a separate detection algorithm. For practical evaluation, we made recordings of speech signals (both loudspeaker-playback and human speaker) with a planar 4-channel prototype array in a medium-sized room. The proposed approach clearly outperforms existing coincident localization methods. We achieve high accuracy (2◦ mean absolute azimuth error at 0 dB SNR) for static sources, while being able to quickly follow rapid source angle changes.
Download Phantom Source Widening With Deterministic Frequency Dependent Time Delays We present a novel method to adjust the perceived width of a phantom source by varying the deterministic inter channel time difference (ICT D) in a pair of signals over frequency. In contrast to given literature that focuses on random phase over frequency, our paper considers a deterministic approach that is open to a more systematic evaluation. Two allpass structures are described, finite impulse response (FIR) and infinite impulse response (IIR), for phase-based phantom source widening and evaluated in a formal listening test. Varying ICT D over frequency essentially alters the inter-aural cross correlation coefficient at the ears of a listener and in this way provides a robust way to control the auditory source width. The subjective evaluation results fully support our observations for both noise and speech signals.
Download Pitch Shifting of Audio Signals Using the Constant-Q Transform Pitch-scale modifications of polyphonic music are usually performed by manipulating the time-frequency representation of the input signal. Most approaches proposed in the past are thereby based on the Fourier transform although its linear frequency bin spacing is known to be inadequate to some degree for analysing and processing music signals. Recently invertible constant-Q transforms (CQT) featuring high Q-factors have been proposed exhibiting a more suitable geometrical bin spacing. In this paper a frequency domain pitch-shifting approach based on the CQT is proposed. The CQT is specifically attractive for pitch-shifting because it can be implemented by frequency translation (shifting partials along the frequency axis) as opposed to spectral stretching in the Fourier transform domain. Furthermore, the high time resolution of CQT at high frequencies improves transient preservation. Audio examples are provided to illustrate the results achieved with the proposed method.