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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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Book
01 Jan 1990
TL;DR: The Fourier series in spectral analysis and function approximation, the Fourier transformation and generalized signals, and some of its applications analog signal processing systems and systems design of digital filters.
Abstract: The Fourier series in spectral analysis and function approximation the Fourier transformation and generalized signals the Laplace transformation and some of its applications analogue signal processing systems digitization of analogue signals discrete signals and systems design of digital filters the fast fourier transform and its applications stochastic signals and power spectra finite word-length effects in digital signal processors linear estimation and adaptive filtering.

281 citations

Patent
28 Jun 2002
TL;DR: In this paper, the authors describe a digital audio player with a menu-driven user interface for selecting, sorting, and playback of stored audio data files, including jazz, pop, and rock.
Abstract: A digital audio player (10) and a method for processing encoded digital audio data, wherein the digital audio data is encoded using one of a plurality of encoding formats. The exemplary audio data player includes a hard disk or other data storage medium (32) for storing data files, a microcontroller (22), buffer memory (25) for anti-skip protection, and an audio decoder (12). The encoded audio data files and associated decoder files are downloaded from a personal computer or similar device to the audio data player hard drive. The player provides a menu-driven user interface (21, 26) for selection, sorting, and playback of stored audio data files. The audio decoder, generally a digital signal processor, provides various preset equalization modes. The preset modes are specific to audio genres such as jazz, pop, and rock. The user may select a specific equalization mode by using the user interface (21, 26), or the preset equalization mode will be automatically set based on the genre, or other attribute information, included in a tag portion of the audio data file.

281 citations

Patent
14 Aug 2001
TL;DR: In this paper, the authors present systems and methods for receiving live speech, converting the speech to text, and transferring the text to a user, as desired, in one or more different languages.
Abstract: The present invention relates to systems and methods for audio processing. For example, the present invention provides systems and methods for receiving live speech, converting the speech to text, and transferring the text to a user. As desired, the speech or text can be translated into one or more different languages. Systems and methods for real-time conversion and transmission of speech and text are provided.

278 citations

Patent
14 May 1999
TL;DR: In this article, an image pickup device (14), an audio pickup device(12), and an audio source locator (10) are used to determine a direction of the audio source relative to a reference point.
Abstract: A system, such as a video conferencing system, is provided which includes an image pickup device (14), an audio pickup device (12), and an audio source locator (10). The image pickup device (14) generates image signals representative of an image, while the audio pickup device (12) generates audio signals representative of sound from audio source, such as speaking person. The audio source locator (10) processes the image signals and audio signals to determine a direction of the audio source relative to a reference point. The system can further determine a location of the audio source relative to the reference point. The reference point can be a camera (14). The system can use the direction or location information to frame a proper camera shot which would include the audio source.

277 citations

Patent
13 Jun 2011
TL;DR: In this paper, a user speech model associated with the user may be accessed and a determination may be made background audio in the audio signal is below a defined threshold, in response to determining that the background audio is below the defined threshold.
Abstract: An audio signal generated by a device based on audio input from a user may be received. The audio signal may include at least a user audio portion that corresponds to one or more user utterances recorded by the device. A user speech model associated with the user may be accessed and a determination may be made background audio in the audio signal is below a defined threshold. In response to determining that the background audio in the audio signal is below the defined threshold, the accessed user speech model may be adapted based on the audio signal to generate an adapted user speech model that models speech characteristics of the user. Noise compensation may be performed on the received audio signal using the adapted user speech model to generate a filtered audio signal with reduced background audio compared to the received audio signal.

276 citations


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