Topic
Audio signal processing
About: Audio signal processing is a research topic. Over the lifetime, 21463 publications have been published within this topic receiving 319597 citations. The topic is also known as: audio processing & Acoustic signal processing.
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Papers
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01 Jan 1997TL;DR: In this paper, a review of algorithms for perceptually transparent coding of high-fidelity (CD-quality) digital audio signals is presented, including algorithms which manipulate transform components and subband signal decompositions.
Abstract: Considerable research has been devoted to the development of algorithms for perceptually transparent coding of high-fidelity (CD-quality) digital audio. As a result, many algorithms have been proposed and several have now become international and/or commercial product standards. This paper reviews algorithms for perceptually transparent coding of CD-quality digital audio, including both research and standardization activities. First, psychoacoustic principles are described with the MPEG psychoacoustic signal analysis model 1 discussed in some detail. Then, we review methodologies which achieve perceptually transparent coding of FM- and CD-quality audio signals, including algorithms which manipulate transform components and subband signal decompositions. The discussion concentrates on architectures and applications of those techniques which utilize psychoacoustic models to exploit efficiently masking characteristics of the human receiver. Several algorithms which have become international and/or commercial standards are also presented, including the ISO/MPEG family and the Dolby AC-3 algorithms. The paper concludes with a brief discussion of future research directions.
77 citations
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IBM1
TL;DR: In this paper, a low resolution, compressed and independent frames derived from the encoded digital video or audio information are used to facilitate the operation of user-requested VTR-like speed change functions associated with digital video and digital audio frames.
Abstract: Systems, methods, and computer products that facilitate transmission of information used for fast and responsive video and audio playback at non-standard, trick mode speeds. An embodiment of the present invention uses low resolution, compressed, and independent frames derived from the encoded digital video or audio information to facilitate the operation of user-requested VTR-like speed change functions associated with digital video and digital audio frames. The present invention greatly simplifies locating specific frames in a video or audio presentation for purposes such as fast forward and fast reverse scanning that is typically used in digital editing.
77 citations
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TL;DR: An algorithm to convert stereo to five-channel sound reproduction is presented and it is revealed that the proposed algorithm is preferred for both on and off the reference listening position (sweet spot).
Abstract: While stereo music reproduction was a dramatic advance over mono, recently a transition to multichannel audio has created a more involving experience for listeners. An algorithm to convert stereo to five-channel sound reproduction is presented. An effective sound distribution to the surround channels is achieved by using a cross-correlation technique, and a robust stereo image is obtained using principal component analysis. Informal listening tests comparing this scheme with other methods revealed that the proposed algorithm is preferred for both on and off the reference listening position (sweet spot).
77 citations
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TL;DR: In this article, an adaptive speech enhancement filter for improving clarity of a voice spoken within an interior cabin having ambient noise includes an adaptive acoustic echo cancellation system for receiving an audio signal and removing the second component in the filtered audio signal to provide an echo-cancelled audio signal, wherein the first adaptation rate and the second adaptation rate are different from each other.
Abstract: A cabin communication system for improving clarity of a voice spoken within an interior cabin having ambient noise includes an adaptive speech enhancement filter for receiving an audio signal that includes a first component indicative of the spoken voice, a second component indicative of a feedback echo of the spoken voice and a third component indicative of the ambient noise, the speech enhancement filter filtering the audio signal by removing the third component to provide a filtered audio signal, the speech enhancement filter adapting to the audio signal at a first adaptation rate, and an adaptive acoustic echo cancellation system for receiving the filtered audio signal and removing the second component in the filtered audio signal to provide an echo-cancelled audio signal, the echo cancellation signal adapting to the filtered audio signal at a second adaption rate, wherein the first adaptation rate and the second adaptation rate are different from each other so that the speech enhancement filter does not adapt in response to operation of the echo-cancellation system and the echo-cancellation system does not adapt in response to operation of the speech enhancement filter.
77 citations
01 Jan 2001
TL;DR: A model for physically based synthesis of collision sounds is proposed, focused on the non-linear contact force, for which both analytical and experimental results are presented.
Abstract: A model for physically based synthesis of collision sounds is proposed. Attention is focused on the non-linear contact force, for which both analytical and experimental results are presented. Numerical implementation of the model is discussed, with regard to accuracy and efficiency issues. As an application, a physically based audio effect is presented.
77 citations