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.

Audio data playback clock signal generating apparatus for use in digital VCR

Abstract: An audio data playback clock signal generating apparatus of a digital VCR synchronizes an audio frame playback synchronization signal with a video frame synchronization signal when an audio data recorded in a magnetic tape is played back based on a phase error of an audio data frame size AF-SIZE and an audio data playback clock signal of the previous video frame. The audio data playback clock signal generating apparatus comprises an audio data frame size decoder, a low pass filter, a voltage-controlled oscillator, and phase error generating block and can be implemented by a large scale integration(LSI) to thereby provide a non-complex circuit and an accurate synchronization between the audio frame playback synchronization signal and the video frame synchronization signal.

Audio intrusion detection system

Abstract: A supervisory circuit for use with an audio intrusion detection system is disclosed. The supervisory circuit periodically generates an audio test signal which is supplied to a sounder which emits an audio test sound. The audio test sound is directed into a volume of space, in the same volume of space as which the audio intrusion detection system is directed to detect. The audio intrusion detection system detects the test sound and generates an audio test signal in response thereto. During the generation of the audio test sound, the comparing apparatus of the audio intrusion detection system is disabled. The audio test signal generated by the audio intrusion detection system is then compared to a test threshold signal. A test result signal is generated in response to the comparison with the test result signal indicative of the operability of the audio intrusion detection system.

AudioCubes: a distributed cube tangible interface based on interaction range for sound design

Abstract: AudioCubes is a novel tangible user interface allowing any person interested by sound design such as sound creators, and music trainers to intuitively explore and create dynamically changing sound. A new sound is created by manipulating distributed cube tangible user interface that can be coupled wirelessly by locating them in the interaction range of each other on a table. At any time, a sound processing network combines operational properties of AudioCubes, such as location on a plane or in space, movement, arrangement with other cubes, and layout. Sound algorithm parameters and the configuration of the sound processing network can be changed simultaneously, allowing a fast and convenient exploration of sound creation space that creates a new interaction technique for creating sounds.

Hearing aid, and hearing-aid processing method and integrated circuit for hearing aid

Abstract: PROBLEM TO BE SOLVED: To convey the user's personal intention to a hearing aid using a method that does not place heavy physical and psychological loads. SOLUTION: The present invention relates to a hearing aid 10 to be worn by a user for auditory compensation of the user, comprising: at least one microphone 11, 12 which converts a sound to an input signal; a hearing-aid signal processing unit 15 configured to generate an output signal from the input signal; a receiver 16 which outputs, as a sound, the output signal generated by the hearing-aid signal processing unit 15; and a hearing-aid processing control unit 14 configured to generate control information for controlling signal processing, on the basis of a non-audible sound which is made by the user and is hard to hear from outside; wherein, when the hearing-aid processing control unit 14 generates the control information, the hearing-aid signal processing unit 15 is configured to generate the output signal according to the generated control information. COPYRIGHT: (C)2010,JPO&INPIT

Hearing system beamformer

Abstract: The present invention, generally speaking, picks up a voice or other sound signal of interest and creates a higher voice-to-background-noise ratio in the output signal so that a user enjoys higher intelligibility of the voice signal. In particular, beamforming techniques are used to provide optimized signals to the user for further increasing the understanding of speech in noisy environments and for reducing user listening fatigue. In one embodiment, signal-to-noise performance is optimized even if some of the binaural cues are sacrificed. In this embodiment, an optimum mix ratio or weighting ratio is determined in accordance with the ratio of noise power in the binaural signals. Enhancement circuitry is easily implemented in either analog or digital form and is compatible with existing sound processing methods, e.g., noise reduction algorithms and compression/expansion processing. The sound enhancement approach is compatible with, and additive to, any microphone directionality or noise cancelling technology.