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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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Journal ArticleDOI
TL;DR: A heuristic rule-based procedure is proposed to segment and classify audio signals and built upon morphological and statistical analysis of the time-varying functions of these audio features.
Abstract: While current approaches for audiovisual data segmentation and classification are mostly focused on visual cues, audio signals may actually play a more important role in content parsing for many applications. An approach to automatic segmentation and classification of audiovisual data based on audio content analysis is proposed. The audio signal from movies or TV programs is segmented and classified into basic types such as speech, music, song, environmental sound, speech with music background, environmental sound with music background, silence, etc. Simple audio features including the energy function, the average zero-crossing rate, the fundamental frequency, and the spectral peak tracks are extracted to ensure the feasibility of real-time processing. A heuristic rule-based procedure is proposed to segment and classify audio signals and built upon morphological and statistical analysis of the time-varying functions of these audio features. Experimental results show that the proposed scheme achieves an accuracy rate of more than 90% in audio classification.

473 citations

Patent
05 Jun 2006
TL;DR: In this article, a real-time audio-on-demand communication system is proposed, which provides realtime playback of audio data transferred via telephone lines or other communication links. But, the system is not suitable for multimedia applications.
Abstract: An audio-on-demand communication system provides real-time playback of audio data transferred via telephone lines or other communication links. One or more audio servers include memory banks which store compressed audio data. At the request of a user at a subscriber PC, an audio server transmits the compressed audio data over the communication link to the subscriber PC. The subscriber PC receives and decompresses the transmitted audio data in less than real-time using only the processing power of the CPU within the subscriber PC. According to one aspect of the present invention, high quality audio data compressed according to lossless compression techniques is transmitted together with normal quality audio data. According to another aspect of the present invention, metadata, or extra data, such as text, captions, still images, etc., is transmitted with audio data and is simultaneously displayed with corresponding audio data. The audio-on-demand system also provides a table of contents indicating significant divisions in the audio clip to be played and allows the user immediate access to audio data at the listed divisions. According to a further aspect of the present invention, servers and subscriber PCs are dynamically allocated based upon geographic location to provide the highest possible quality in the communication link.

470 citations

Journal ArticleDOI
TL;DR: This paper, Part II, generalizes the basic BCC schemes presented in Part I and includes BCC for multichannel signals and employs an enhanced set of perceptual spatial cues for BCC synthesis.
Abstract: Binaural Cue Coding (BCC) is a method for multichannel spatial rendering based on one down-mixed audio channel and side information. The companion paper (Part I) covers the psychoacoustic fundamentals of this method and outlines principles for the design of BCC schemes. The BCC analysis and synthesis methods of Part I are motivated and presented in the framework of stereophonic audio coding. This paper, Part II, generalizes the basic BCC schemes presented in Part I. It includes BCC for multichannel signals and employs an enhanced set of perceptual spatial cues for BCC synthesis. A scheme for multichannel audio coding is presented. Moreover, a modified scheme is derived that allows flexible rendering of the spatial image at the receiver supporting dynamic control. All aspects of complete BCC encoder and decoder implementations are discussed, such as down-mixing of the input signals, low complexity estimation of the spatial cues, and quantization and coding of the side information. Application examples are given and the performance of the coder implementations are evaluated and discussed based on subjective listening test results.

464 citations

Patent
25 Feb 1997
TL;DR: In this paper, a communication system for simulataneously transmitting ancillary codes and audio signals via a conventional audio communications channel using perceptual coding techniques is described, where an encoder monitors an audio channel to detect "opportunities" to insert an anciliary code such that the inserted signals are masked by the audio signal.
Abstract: A communication system for simulataneously transmitting ancillary codes and audio signals via a conventional audio communications channel using perceptual coding techniques is disclosed. An encoder monitors an audio channel to detect 'opportunities' to insert an ancillary code such that the inserted signals are masked by the audio signal, as defined by the 'perceptual entropy envelope' of the audio signal. An ancillary code containing, for example, an ID or serial number, is encoded as one or more whitened spread stpectrum signals and/or a narrowband FSK ancillary code and transmitted at a time, frequency and/or level such that the data signal is masked by the audio signal. A decoder at a receiving location recovers the encoded ID or serial number.

459 citations

Journal ArticleDOI
TL;DR: This paper proposes to analyze a large number of established and recent techniques according to four transverse axes: 1) the acoustic impulse response model, 2) the spatial filter design criterion, 3) the parameter estimation algorithm, and 4) optional postfiltering.
Abstract: Speech enhancement and separation are core problems in audio signal processing, with commercial applications in devices as diverse as mobile phones, conference call systems, hands-free systems, or hearing aids. In addition, they are crucial preprocessing steps for noise-robust automatic speech and speaker recognition. Many devices now have two to eight microphones. The enhancement and separation capabilities offered by these multichannel interfaces are usually greater than those of single-channel interfaces. Research in speech enhancement and separation has followed two convergent paths, starting with microphone array processing and blind source separation, respectively. These communities are now strongly interrelated and routinely borrow ideas from each other. Yet, a comprehensive overview of the common foundations and the differences between these approaches is lacking at present. In this paper, we propose to fill this gap by analyzing a large number of established and recent techniques according to four transverse axes: 1 the acoustic impulse response model, 2 the spatial filter design criterion, 3 the parameter estimation algorithm, and 4 optional postfiltering. We conclude this overview paper by providing a list of software and data resources and by discussing perspectives and future trends in the field.

452 citations


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