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.

Automatic sound field correcting device and computer program therefor

Abstract: An automatic sound field correcting device executes a signal process to the plurality of audio signals on respective correspondent signal transmission paths, and outputs them to a plurality of correspondent speakers (106) to correct sound characteristics on the respective signal transmission paths. Namely, a measurement signal (211) is supplied to each signal transmission path, and a measurement sound (250) corresponding to it is outputted from the speaker (106) to a sound space (260). The outputted measurement sound is detected as a detecting signal (205). The frequency characteristic of the audio signal on each signal transmission path is corrected by an equalizer (120), and a gain value of the equalizer (120) is determined by a correction amount determining unit (112). A frequency characteristics correction is performed predetermined times. At a first correction, the correction amount determining unit (112) determines the correction amount (210) by performing a frequency analysis, based on the detecting signal (205) corresponding to the measurement sound (250) actually outputted to the sound space (260). On the contrary, at and after a second correction, the correction amount determining unit (112) determines the correction amount (210) based on the detecting signal (205) or an output signal (201) of the equalizer (120). Namely, at and after the second correction, the output signal (201) of the equalizer (120) is supplied to the correction amount determining unit (112) in a signal processing circuit (102) as the need arises, and the frequency characteristics correction is performed without actually outputting the measurement sound (250) to the sound space (206).

Intelligent systems for mixing multichannel audio

Abstract: Multichannel signal processing techniques are usually concerned with extracting information about sources from several received signals. In this paper, we describe an emerging field of multichannel audio signal processing where the inter-channel relationships are exploited in order to manipulate the multichannel content. Applications to real-time, automatic audio production are described and the necessary technologies and the architecture of such systems are presented. The current state of the art is reviewed, and directions of future research are also discussed.

Signal modification method for efficient coding of speech signals

Abstract: For determining a long-term-prediction delay parameter characterizing a long term prediction in a technique using signal modification for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, a feature of the sound signal is located in a previous frame, a corresponding feature of the sound signal is located in a current frame, and the long-term-prediction delay parameter is determined for the current frame while mapping, with the long term prediction, the signal feature of the previous frame with the corresponding signal feature of the current frame. In a signal modification method for implementation into a technique for digitally encoding a sound signal, the sound signal is divided into a series of successive frames, each frame of the sound signal is partitioned into a plurality of signal segments, and at least a part of the signal segments of the frame are warped while constraining the warped signal segments inside the frame. For searching pitch pulses in a sound signal, a residual signal is produced by filtering the sound signal through a linear prediction analysis filter, a weighted sound signal is produced by processing the sound signal through a weighting filter, the weighted sound signal being indicative of signal periodicity, a synthesized weighted sound signal is produced by filtering a synthesized speech signal produced during a last subframe of a previous frame of the sound signal through the weighting filter, a last pitch pulse of the sound signal of the previous frame is located from the residual signal, a pitch pulse prototype of given length is extracted around the position of the last pitch pulse of the sound signal of the previous frame using the synthesized weighted sound signal, and the pitch pulses are located in a current frame using the pitch pulse prototype.

System and method for automated audio/video archive and distribution

Abstract: A digital audio/video archive and distribution system (10) and method utilize a signal capture and compression encoding subsystem (12) to translate an analog signal from at least one media source (14) into at least one digital signal segment. The digital signal segment is automatically correlated with identifying information (18) input via an Internet connection, and both are stored in a searchable database subsystem (24). Upon a user request (28) received via an Internet connection, or programming prompt, a graphical user interface (GUI), such as a web-based user interface, is created to distribute one or more of the stored digital signal segments.

Teminology in digital signal processing

Abstract: The committee on Digital Signal Processing of the IEEE Group on Audio and Electroacoustics has undertaken the project of recommending terminology for use in papers and texts on digital signal processing. The reasons for this project are twofold. First, the meanings of many terms that are commonly used differ from one author to another. Second, there are many terms that one would like to have defined for which no standard term currently exists. It is the purpose of this paper to propose terminology which we feel is self-consistent, and which is in reasonably good agreement with current practices. An alphabetic index of terms is included at the end of the paper.