Authors:Wühle, Tom; Merchel, Sebastian; Altinsoy, M. Ercan
Affiliation:Dresden University of Technology, Chair of Acoustic and Haptic Engineering, Dresden, Germany
One solution to realize spatial sound reproduction without a large number of distributed loudspeakers is to create virtual sources in the desired directions by using highly focusing real sources that project sound on reflective boundaries. Auditory events are then ideally located in the direction of the virtual sources created by sound projection rather than the direction of the real sources. However, due to physically limited focusing capabilities of real sources, the perception of the listener is also influenced by sound that is directly radiated from the real source and, therefore, arrives earlier at the position of the listener. This study showed how emerging precedence caused by the leading direct sound affected auditory perception in scenarios with lagging projected sound. The level range from which the direct sound caused first noticeable changes in auditory events until it finally dominated the localization was found to be approximately 20 dB. It was shown that localization dominance of the projected sound did not immediately occur after localization dominance of the direct sound vanished. However, localization dominance of the projected sound occurred even though the presence of the direct sound was still perceptible. The results indicate that the temporal structure of the direct sound plays an important perceptual role in scenarios with projected sound. Real playback signals with complex temporal structures causing impulsive loudness fluctuations were shown to be more favorable for the perceptual dominance of the leading direct sound. Such structures are therefore critical in terms of sound projection.
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Authors:Sanalatii, Maryna; Herzog, Philippe; Guillermin, Régine; Melon, Manuel; Poulain, Nicolas; Le Roux, Jean-Christophe
Affiliation:Laboratoire de Mécanique et d'Acoustique (LMA), Marseille, France; Laboratoire d'Acoustique de l'Université du Mans (LAUM), Le Mans Cedex 9, France; Centre de Transfert de Technologie du Mans (CTTM), Le Mans, France
Although the acoustic radiation of common loudspeaker systems can be easily measured using modern measurement techniques, the process requires free-field conditions that may be difficult to satisfy. Large anechoic rooms are very expensive and outdoor measurements are subject to uncontrollable weather conditions. This paper proposes the use of the radiation mode (RM) method to estimate the frequency response and directivity pattern of loudspeaker systems. The underlying theory and method principle are first described and then assessed in both an anechoic room and large non-anechoic hall by measuring four loudspeaker systems with different radiation patterns. Results show a satisfactory level of accuracy for the proposed method across all sources tested and both measurement rooms, especially when considering the reduced number of measurement points needed. These examples are then complemented with a preliminary parametric study based on the simulation of a tall system, namely a line array for which standard measurement techniques are not applicable. More specifically, the influences of identification point locations, noise, and RM series truncation are all investigated. The output of these simulations illustrates the potential of this method to characterize sound sources that cannot be measured using classical means.
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Affiliation:Université de Toulon / CNRS PRISM / Aix-Marseille Université, Marseille, France
In this study the author describes an algorithm that can discriminate transcoded from true lossless audio files, i.e. a file that was obtained by directly ripping a genuine audio CD from a file that was decoded from a lossy audio format (e.g. MP3, AAC) and re-encoded with a lossless format. This algorithm should allow both consumers and online music dealers to check the authenticity of lossless music files. Although research focuses on the MPEG AAC codec, the approach can also be applied to the MP3 codec. It is based on the detection of quantization errors in the time-frequency domain without using machine learning. Tests with a large database of 1576 audio files, original and transcoded with AAC from iTunes, show null false positive ratios and very low false negative ratios, possibly null for high-precision settings, which is better than other methods. It was shown that a perfectly accurate classification can be reached, provided a sufficiently long computation time is allowed. Furthermore, the method is naturally immune to truncation of audio files and global gain modification.
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Affiliation:Dolby Laboratories, San Francisco, CA, USA
This study shows that PESQ can be used as a tool to evaluate degradations from listener echo and duplex impairments caused by echo-mitigation algorithms such as echo cancellation or echo suppression. Both the PESQ-based metric and standards 3GPP TS 26.132 and P.502 share the approach of testing with real speech and comparing an impaired signal to an unimpaired reference. However, unlike 3GPP/P.502, PESQ provides tools for accurate time alignment of the signals that function even with temporally varying delay (jitter) and thus allow measurement in IP-based networks. Moreover, the PESQ metric follows the common practice of calculating PESQ values for any test condition with several speech samples, which stabilizes the quality estimate. In contrast 3GPP prescribes the use of a single test signal, which causes potentially misleading sampling error. Finally, the well-developed perceptual model underlying PESQ generates a perceptually relevant one-dimensional result. This is suitable for benchmark or regression testing. In contrast 3GPP and P.502 use only rudimentary perceptual models or no models at all and generate multidimensional results that are unwieldy when used for performance comparison or tracking
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Setting up and successfully operating an audio network is not something to be taken lightly if you want it to be reliable. Higher-level protocols that handle discovery, connection management, and network redundancy are available. While one might want all of these things to be handled “under one roof” it seems common for a variety of tools and solutions to have to be layered in order to create a system that functions. While there are a number of common basic features to network protocols for real time media, the precise ways in which they are implemented may differ between systems. Networked audio sessions from the 145th Convention are summarized.
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