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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.


Papers
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BookDOI
17 Sep 2008
TL;DR: In this paper, the state of the art in important areas of speech and audio signal processing is discussed, including multi-microphone systems, specific approaches for noise reduction, and evaluations of speech signals and speech processing systems.
Abstract: The book reflects the state of the art in important areas of speech and audio signal processing. It presents topics which are missed so far and most recent findings in the field. Leading international experts report on their field of work and their new results. Considerable amount of space is covered by multi-microphone systems, specific approaches for noise reduction, and evaluations of speech signals and speech processing systems. Multi-microphone systems include automatic calibration of microphones, localisation of sound sources, and source separation procedures. Also covered are recent approaches to the problem of adaptive echo and noise suppression. A novel solution allows the design of filter banks exhibiting bands spaced according to the Bark scale und especially short delay times. Furthermore, a method for engine noise reduction and proposals for improving the signal/noise ratio based on partial signal reconstruction or using a noise reference are reported. A number of contributions deal with speech quality. Besides basic considerations for quality evaluation specific methods for bandwidth extension of telephone speech are described. Procedures to reduce the reverberation of audio signals can help to increase speech intelligibility and speech recognition rates. In addition, solutions for specific applications in speech and audio signal processing are reported including, e.g., the enhancement of audio signal reproduction in automobiles and the automatic evaluation of hands-free systems and hearing aids.

68 citations

Patent
05 Apr 2011
TL;DR: In this article, Delay Matching (synchronization) in the audio/video system comprises adjusting the audio path delay to be substantially equal to the video path delay, which is referred to as delay matching.
Abstract: An audio/video system comprises an audio signal processing path having an audio path delay and a video signal processing path having a video path delay. The audio path delay may be different from the video path delay. The audio path delay and/or the video path delay may change, for example because of replacement of a component within the audio signal processing path or the video signal processing path. Delay matching (synchronization) in the audio/video system comprises adjusting the audio path delay to be substantially equal to the video path delay. Matching the audio path delay to the video path delay generally includes adding delay to the signal processing path with the lesser delay.

68 citations

BookDOI
09 Oct 2001
TL;DR: In this paper, the authors provide a complete treatment on the theoretical and practical aspects of synchronization and channel estimation from the standpoint of digital signal processing, focusing on the systematic approach to algorithm development, and the linked algorithm-architecture methodology in digital receiver design.
Abstract: From the Publisher: Digital Communication Receivers offers a complete treatment on the theoretical and practical aspects of synchronization and channel estimation from the standpoint of digital signal processing. The focus on these increasingly important topics, the systematic approach to algorithm development, and the linked algorithm-architecture methodology in digital receiver design are unique features of this book. The material is structured according to different classes of transmission channels. In Part C, baseband transmission over wire or optical fiber is addressed. Part D covers passband transmission over satellite or terrestrial wireless channels. Part E deals with transmission over fading channels. Designed for the practicing communication engineer and the graduate student, the book places considerable emphasis on helpful examples, summaries, illustrations, and bibliographies. Contents include basic material, baseband communications, passband transmission, receiver structure for PAM signals, synthesis of synchronization algorithms, performance analysis of synchronizers, bit error degradation caused by random tracking errors, frequency estimation, timing adjustment by interpolation, DSP system implementation, characterization, modeling, and simulation of linear fading channels, detection and parameter synchronization on fading channels, receiver structures for fading channels, parameter synchronization for flat fading channels, and parameter synchronization for selective fading channels.

68 citations

Patent
12 Jul 2010
TL;DR: In this article, an adaptive gain control system and related operating method for digital audio samples is presented for use with a digital media encoding system that transmits encoded media streams to a remotely-located presentation device such as a media player.
Abstract: An adaptive gain control system and related operating method for digital audio samples is provided. The method is suitable for use with a digital media encoding system that transmits encoded media streams to a remotely-located presentation device such as a media player. The method begins by initializing the processing of a media stream. Then, the method adjusts the gain of a first set of digital audio samples in the media stream using a fast gain adaptation scheme, resulting in a first group of gain-adjusted digital audio samples having normalized volume during presentation. The method continues by adjusting the gain of a second set of digital audio samples in the media stream using a steady state gain adaptation scheme that is different than the fast gain adaptation scheme, resulting in a second group of gain-adjusted digital audio samples having normalized volume during presentation.

67 citations

Patent
11 Dec 2009
TL;DR: In this paper, a method for coding a multi-channel audio signal representing a sound scene comprising a plurality of sound sources is provided, which comprises decomposing the multichannel signal into frequency bands and, per frequency band, obtaining directivity information per sound source of the sound scene.
Abstract: A method for coding a multi-channel audio signal representing a sound scene comprising a plurality of sound sources is provided. This method comprises decomposing the multi-channel signal into frequency bands and, per frequency band, obtaining directivity information per sound source of the sound scene, the information being representative of the spatial distribution of the sound source in the sound scene, of selecting a set of sound sources of the sound scene constituting principal sources, of matrixing the selected principal sources to obtain a sum signal with a reduced number of channels and, of coding the directivity information and of forming a binary stream comprising the coded directivity information, the binary stream being transmittable in parallel with the sum signal. A decoding method is also provided that is able to decode the sum signal and the directivity information to obtain a multi-channel signal, to an adapted coder and an adapted decoder.

67 citations


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Performance
Metrics
No. of papers in the topic in previous years
YearPapers
202319
202263
2021217
2020525
2019659
2018597