Aachener Beiträge zu digitalen Nachrichtensystemen – serie
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Del 39 - Aachener Beiträge zu digitalen Nachrichtensystemen
Multi Channel Audio Processing: Enhancement, Compression and Evaluation of Quality
Häftad, Engelska, 2014
550 kr
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The demand for high definition audio and video services is rapidly increasing. Two representative examples for this are audio-visual conferencing or video storage and delivery. In this context, efficient techniques are required for enhancement and compression of multi channel audio signals with compatibility to, e.g., mono or stereo systems. In this thesis, novel signal processing algorithms for both enhancement and compression of multi channel signals are developed and theoretical performance bounds are derived. Additionally, a novel instrumental quality measure for the evaluation of multi channel signal processing algorithms is proposed.Enhancement schemes for both the recording and the reproduction side are introduced. This includes the optimization of a near field filter-and-sum beamformer to achieve a target directivity characteristic at the recording side. For the reproduction side, an efficient postfilter is presented which increases the speech intelligibility by taking the positive influence of early room reflections into account.The main part of this thesis covers multi channel predictive compression of audio signals. A predictive multi channel coding system is presented and analyzed. Performance bounds are derived and two methods for an adaptive bit rate distribution between inter channel and intra channel prediction are devised. Novel multi channel noise shaping concepts are introduced. The performance of the compression system is quantified by instrumental measures.A novel instrumental measure is introduced for the evaluation of multi channel signal enhancement and compression. It combines the proven single channel quality measure PEAQ with a binaural auditory model and a mathematical model of cognitive behavior, providing a reliable evaluation of quality perception and spatial fidelity. The inclusion of spatial information into the instrumental quality measurement leads to a consistently high correlation between the instrumental measure and a listening test.
Del 48 - Aachener Beiträge zu digitalen Nachrichtensystemen
Active Noise and Occlusion Effect Cancellation in Headphones and Hearing Aids
Häftad, Engelska, 2010
553 kr
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The perception of one’s own voice is distorted when telephoning with headsets, or wearing hearing aids. The reason for this is the so-called occlusion effect, which occurs when ear canals are completely or partially closed by the headset or hearing aid. The occlusion causes amplification at low frequencies, and attenuation at high frequencies of one’s own voice. The unnatural perception of one’s own voice and of noise caused by chewing and swallowing are among the most common complaints of users. Furthermore, environmental noise might impair perception. In this thesis, both the unnatural perception of one’s own voice and the disturbance by environmental noise are tackled by a novel signal processing approach. The proposed solution solves the problem of the occlusion effect by actively emitting a compensation signal through the integrated loudspeaker. The approach is called Occlusion Effect Cancellation (OEC) and significantly improves the perception of one’s own voice and of the acoustic environment. This novel approach combines methods of active noise cancellation (ANC, Noise Cancelling Headphone) with a personalized design. The bilateral headset contains two additional microphones per side, one inner and one outer, to acquire signals for the calculation of the compensation signals. A correctly balanced processing of the two microphone signals results in a "digital ear opening" and a much more natural perception of both one’s own voice and of the environment. The extent of the digital ear opening is controllable. The system can also be operated as a noise cancelling headphone by changing the parameters to a conventional design to create an acoustic isolation from the environment.This thesis proposes a novel robust approach based on digital filtering to solve the described problems. A combination of feedback and feedforward filter design allows for either approaching personal silence or a natural perception of one’s own voice and the acoustic environment.