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.

Architectures for Digital Signal Processing

Abstract: From the Publisher: Digital Signal Processing is a rapidly expanding area for evaluation and development of efficient measures for representation, transformation and manipulation of signals. This book fuses signal processing algorithms and VLSI circuit design to assist digital signal processing architecture developers. The author also shows how this technique can be used in applications such as: signal transmission and storage, manufacturing process quality control and assurance, autonomous mobile system control and biomedical process analysis.

Robustness and Regularization of Personal Audio Systems

Abstract: As well as being able to reproduce sound in one region of space, it would be useful to reduce the level of reproduced sound in other spatial regions, with a “personal audio” system. For mobile devices this is motivated by issues of privacy for the user and the need to reduce annoyance for other people nearby. Such personal audio systems can be realized with arrays of loudspeakers that become superdirectional at low frequencies, when the array dimensions are small compared with the acoustic wavelength. The design of the array then becomes a compromise between performance and array effort, defined as the sum of mean squared driving signals. Various methods of formulating this tradeoff as a regularization problem have been suggested and the connection between these formulations is discussed. Large array efforts are due to strongly self-cancelling multipole arrays. A concern is then the robustness of such an array to variations in the acoustic environment and driver sensitivity and position. The design of an array that is robust to these uncertainties then leads to a generalization of regularization.

Method and apparatus for accommodating primary content audio and secondary content remaining audio capability in the digital audio production process

Abstract: The invention enables the inclusion of voice and remaining audio information at different parts of the audio production process. In particular, the invention embodies special techniques for VRA-capable digital mastering and accommodation of VRA by those classes of audio compression formats that sustain less losses of audio data as compared to any codecs that sustain comparable net losses equal or greater than the AC3 compression format. The invention facilitates an end-listener's voice-to-remaining audio (VRA) adjustment upon the playback of digital audio media formats by focusing on new configurations of multiple parts of the entire digital audio system, thereby enabling a new technique intended to benefit audio end-users (end-listeners) who wish to control the ratio of the primary vocal/dialog content of an audio program relative to the remaining portion of the audio content in that program.

Audio device and method of operation therefor

Abstract: An audio device is arranged to present a plurality of test audio signals to a user where each test audio signal comprises a signal component and a noise component. A user preference processor (109) receives user preference feedback for the test audio signals and generates a personalization parameter for the user in response to the user preference feedback and a noise parameter for the noise component of at least one of the test audio signals. An audio processor (113) then processes an audio signal in response to the personalization parameter and the resulting signal is presented to the user. The invention may allow improved characterization of a user thereby resulting in improved adaptation of the processing and thus an improved personalization of the presented signal. The invention may e.g. be beneficial for hearing aids for hearing impaired users.

A method for indicating the presence of speech in an audio signal

Abstract: A voice operated switch employs digital signal processing techniques to examine audio signal frames having harmonic content to identify voiced phonemes and to determined whether the signal frame contains primarily speech or noise. The method and apparatus employ a multiple-stage, delayed-decision adaptive digital signal processing algorithm implemented through the use of commonly available electronic circuit components. Specifically the method and apparatus comprise a plurality of stages, including (1) a low-pass filter to limit examination of input signals to below about one kHz, (2) a digital center-clipped autocorrelation processor whih recognizes that the presence of periodic components of the input signal below and above a peak-related threshold identifies a frame as containing speech or noise, and (3) a nonlinear filtering processor which includes nonlinear smoothing of the frame-level decisions and incorporates a delay, and further incorporates a forward and backward decision extension at the speech-segment level of several tenths of milliseconds to determine whether adjacent frames are primarily speech or primarily noise.