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.

Modulation-scale analysis for content identification

Abstract: For nonstationary signal classification, e.g., speech or music, features are traditionally extracted from a time-shifted, yet short data window. For many applications, these short-term features do not efficiently capture or represent longer term signal variation. Partially motivated by human audition, we overcome the deficiencies of short-term features by employing modulation-scale analysis for long-term feature analysis. Our analysis, which uses time-frequency theory integrated with psychoacoustic results on modulation frequency perception, not only contains short-term information about the signals, but also provides long-term information representing patterns of time variation. This paper describes these features and their normalization. We demonstrate the effectiveness of our long-term features over conventional short-term features in content-based audio identification. A simulated study using a large data set, including nearly 10 000 songs and requiring over a billion audio pairwise comparisons, shows that modulation-scale features improves content identification accuracy substantially, especially when time and frequency distortions are imposed.

Integration of audio and visual information for content-based video segmentation

Abstract: A video sequence usually consists of separate scenes, and each scene includes many shots. For video understanding purposes, it is most important to detect scene breaks. To analyze the content of each scene, detection of shot breaks is also required. Usually, a scene break is associated with a simultaneous change of image, motion, and audio characteristics, while a shot break is only accompanied with changes in image or motion or both. We propose to use audio information along with image and motion information to accomplish segmentation at different levels. Promising results have been obtained with videos digitized from TV programs.

Method for encoding and decoding object-based audio signal and apparatus thereof

Abstract: The present invention relates to a method and apparatus for encoding and decoding object- based audio signals. This audio decoding method includes extracting a first audio signal and a first audio parameter in which a music object are encoded on a channel basis and a second audio signal and a second audio parameter in which a vocal object are encoded on an object basis, from an audio signal, generating a third audio signal by employing at least one of the first and second audio signals, and generating a multi-channel audio signal by employing at least one of the first and second audio parameters and the third audio signal. Accordingly, the amount of calculation in encoding and decoding processes and the size of a bitstream that is encoded can be reduced efficiently.

Programmable auditory prosthesis with trainable automatic adaptation to acoustic conditions

Abstract: An auditory prosthesis (30) comprising a microphone (27) for receiving the sound and producing a microphone signal corresponding to the received sound, an output device for providing audio signals in a form receivable by a user of the prosthesis (30), a sound processing unit (33) operable to receive the microphone signal and carry out a processing operation on the microphone signal to produce an output signal in a form suitable to operate the output device, wherein the sound processing unit (33) is operable in a first mode in which the processing operation comprises at least one variable processing factor which is adjustable by a user to a setting which causes the output signal of the sound processing unit (33) to be adjusted according to the preference of the user for the characteristics of the current acoustic environment.

Sound-processing strategy for cochlear implants

Abstract: A sound processing method for auditory prostheses, such as cochlear implants, which is adapted to improve the perception of loudness by users, and to improve speech perception. The overall contribution of stimuli to simulated loudness is compared with an estimate of acoustic loudness for a normally hearing listener based on the input sound signal. A weighting is applied to the filter channels to emphasize those frequencies which are most important to speech perception for normal hearing listeners when selecting channels as a basis for stimulation.