Abstract: Improved approaches to matching sensitivities of microphones in multi-microphone directional processing systems. These approaches operate to adaptively match microphone sensitivities so that directional noise suppression is robust. As a result, microphone sensitivities remain matched not only over time but also while in actual use. These approaches are particularly useful for hearing aid applications in which directional noise suppression is important.

Abstract: Pathological binaural phase time delay (BPTD) asynchrony is measured at a variety of frequencies and to speech stimuli to develop a BPTD profile for a subject. Then a corrective device (600, 1000) is designed to apply clinical BPTD to compensate for the subject's pathological BPTD. An electronic device (500) is used to measure the subject's ability to comprehend words at a variety of relative time delays between ears to estimate the ideal overall relative time delay. The optimal relative phase shift at a variety of frequencies is also measured. An electronic device (600) may be used to correct the pathological BPTD by delaying sound in different frequency bands differently to the target ear, according to the BPTD profile, or a passive filtered earplug (1000) may be used to correct smaller amounts of BPTD.

Abstract: A transparent panel-form loudspeaker consists of a transparent sound radiation panel that can radiate sound with desired pressure level over a specific frequency range when subjected to the flexural vibration induced by a preselected number of transducers located at specific positions on the peripheral edge of the transparent sound radiation panel and a rigid frame carrying a flexible suspension device which supports the periphery of the transparent sound radiation panel. The transparent sound radiation panel is made of a kind of transparent materials with the ratio of elastic modulus to density in the range from 3 to 180 GPa/(g/cm3) and the ratio of length to thickness of the transparent sound radiation panel in the range from 80 to 600.

Abstract: A system adjusts audio levels so as to maintain a constant perceived audio level at a user's location. The system includes a sensor (110) and an audio device (120). The sensor (110) receives a first audio signal, receives at least one data packet (1000) comprising a second audio signal, determines whether a difference between an average volume level of the first audio signal and the second audio signal exceeds a threshold value, generates a response data packet (1200) including a volume adjustment command when the difference between the average volume level of the first audio signal and the second audio signal exceeds the threshold value, and transmits the response data packet (1200). An audio device (120) transmits the first audio signal, transmits the at least one data packet (1000) to the sensor (110), receives the response data packet (1200), and adjusts an audio level based on the response data packet (1200).

Abstract: A method and system are disclosed for controlling pop noises in a sound broadcasting system. After controllably connecting an output of a drive amplifier to a first predetermined low voltage level through a first switch, a first portion of an operation control data set is input to a digital-to-analog converter (DAC) circuit for driving an output thereof to a second predetermined low voltage level, and a second portion of the operation control data set is also input to the DAC circuit and further to the drive amplifier to bring the output of the drive amplifier to a common mode voltage level over a predetermined rise-up time period for controlling the pop noises.

Abstract: The present invention provides a cancellation signal which may be combined with a radio or television program signal to provide a mix minus signal for use in IFB communication with the talent. A cancellation signal circuit is responsive to a talent signal to delay and gain adjust the talent signal to provide the cancellation signal. The amount of gain and delay necessary may be adjusted by an operator, or automatically determined or set in response to the program signal or the mix minus signal. Automatic operation is provided even in the presence of changing delay and amplitude of the program signal or the talent signal component thereof.

Abstract: The bass compensation device of the present invention includes a high frequency removing circuit for removing a high frequency component of an input sound signal; an absolute value circuit for full-wave rectifying an output signal of the high frequency removing circuit; a band limit circuit for selectively passing only a predetermined frequency range component from the output signal of the absolute value circuit; and an amplifier for amplifying the output signal of the band limit circuit, and bass compensation is performed by the output signal of the amplifier. This structure can realize a bass compensation device excellent in distortion-factor property and transient property.

Abstract: In correcting the sound field, the loudspeakers 6FL to 6WF are sounded by the noise. The attenuation factors of the inter-band attenuators ATF11 to ATFki for adjusting gains of the band-pass filters BPF11 to BPFki to the frequency in respective channels are corrected based on detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. Then, the attenuation factors of the channel-to-channel attenuators ATG1 to ATG5 are corrected based on the detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. Then, the delay times of the delay circuits DLY1 to DLY5 are corrected based on the detection results of the reproduced sounds of the loudspeakers 6FL to 6WF. Then, the attenuation factor of the channel-to-channel attenuator ATGk is corrected based on the detection result of the reproduced sound of the loudspeaker 6WF as the subwoofer, whereby the levels of the reproduced sounds reproduced by the loudspeakers 6FL to 6WF are adjusted to be made flat over the audio frequency band.

Abstract: A speaker apparatus can ensure a large size and area of a diaphragm so as to improve reproducing capability in a low sound range and increase an output sound pressure, and can constitute a plurality of vibrating points (signal control points). An input signal Vin having a plurality of independent channels is inputted, then the signal input Vin is processed in a sound signal processing portion by calculating and adding of an interference canceling signal between the signal control points, by calculating and adding of a sound interference signal for causing the interference between outputs from the signal control points in an arbitrary point, etc., so as to be inputted to transducers that are attached to a single diaphragm. The transducer transduces an electric signal into mechanical vibration. A plurality of the signal control points are generated on the single diaphragm, and each signal control point cause the single diaphragm to vibrate.

Abstract: An audio user interface is provided in which items are represented in an audio field by corresponding synthesized sound sources from where sounds related to the items appear to emanate. At least some of the sound sources are associated in a collection as members of the collection; the collection itself has an associated collection-representing sound source. The collection can be changed in either direction between an un-collapsed state in which the member sound sources are present un-muted in the audio field, and a collapsed state in which the member sound sources are muted with the collection-representing sound source providing an audible presence for the collection in the audio field.

Abstract: A filter coefficient of a graphic equalizer GEQ is corrected based on detection results of reproduced sounds generated by supplying a noise to all frequency band loudspeakers 6FL to 6RR and a low frequency band exclusively reproducing loudspeaker 6WF via the graphic equalizer GEQ. Then, attenuation factors of channel-to-channel attenuators ATG1 to ATG5 are corrected based on the detection results of the reproduced sounds generated by supplying the noise to the loudspeakers 6FL to 6RR via the graphic equalizer. Then, delay times of delay circuits DLY1 to DLYk are corrected based on the detection results of the reproduced sounds generated by supplying the noise to the loudspeakers 6FL to 6WF via the graphic equalizer.

Abstract: A button apparatus with a speaker (100) has a frame (104), a speaker (108), a flexible PCB (110), a metal dome (114), a rubber layer (116), and an operating button (120,124). The frame has a cavity (106) and the speaker is securely received in the cavity. The flexible PCB is located on the frame. The metal dome is located on the flexible PCB, and adapted to electrically connect with the flexible PCB. The rubber layer is located on the flexible PCB. The rubber layer has a protrusion (118) to press the metal dome so that the metal dome is electrically coupling to the flexible PCB. The operating button is located on the rubber layer. The button is adapted to press the protrusion, which in turn presses the metal dome.

Abstract: A noise suppressor device for attaining perceptually preferable noise suppression is disclosed. The device minimizes reduction in quality even in the presence of increased noises. The device is adaptable for use in voice communications systems and speech recognition systems employed in a variety of kinds of noisy environments. The device includes a spectrum subtracter and a spectrum amplitude suppressor that operate on the basis of perceptual weights.

Abstract: A low cost, reliable and accurate audio frequency phase detection system and method includes an audio signal source generating an audio frequency test signal and a phase detector detecting phase reversals in the audio frequency test signal. The signal source may be an active, real time signal generator or a recorded media player playing back a prerecorded signal. The test signal is provided to a user system while the phase detector determines whether or not a phase reversal has occurred in the test signal between the input and output of the user system and illuminates a green or red LED to indicate whether or not a phase reversal has occurred. The test signal includes a lower frequency signal component and a higher frequency signal component that provides a marker for testing the phase of the lower frequency signal component.

Abstract: A plurality of audio inputs which require gain applied are coupled together, the gain is applied and that weighted signal is summed with the output of similar circuits running with different gains. For typical overall gain requirements of 0 to +22.5 dB to be applied in 1.5 dB increments, the new circuit reduces by over two-thirds the number of analog switches required in the level adjusting portion of an audio mixer. More generally, the complexity of the new circuit, in terms of the number of analog switching elements is 2m(1+log2n) for m channels with n gain steps/channel. The number of stages of differing gain is reduced from 2m variable gain amplifiers to 2(1+log2n) fixed gain amplifiers.

Abstract: An entertainment and pacification system for use with a child car seat has speakers mounted in the child car seat with a plurality of audio sources and an anti-noise audio system coupled to the child car seat. A controllable switching system provides for, at any given time, the selective activation of i) one of the audio sources such that the audio signal generated thereby is coupled to one or more of the speakers, and ii) the anti-noise audio system such that an ambient-noise-canceling audio signal generated thereby is coupled to one or more of the speakers. The controllable switching system can receive commands generated at one of first controls located at the child car seat and second controls located remotely with respect to the child car seat with commands generated by the second controls overriding commands generated by the first controls.

Abstract: The present invention relates to an auto loudness circuit for performing loudness compensation automatically depending on the signal level. When the signal level decreases, loudness compensation is slowly introduced and as the signal level increases, loudness compensation is quickly removed. To do so, the auto loudness circuit utilizes a filter circuit with the characteristic of a first order bass boost. The filter circuit maintains a corner frequency which is proportional to the inverse of audio level in order to mimic the Fletcher-Munson curves. Because the circuit employs a capacitance-multiplier with a first order resistance capacitance filter, the bass boost is inversely proportional to the signal level. Thus, bass boost is achieved automatically as the program content changes so that the listener is unaware of significant changes in program material as signal levels change either through increase or decrease in volume, crescendo or new material.

Abstract: The present invention relates to an encoding device for saving the number of bits of codes. In step S11, the differential value between a normalization coefficient Bi to be encoded and a normalization coefficient Bi-1 for an encoding unit Ai-1 in a band adjacent to the lower side of an encoding unit Ai corresponding to the normalization coefficient Bi is computed. In step S12, reference is made to a table in which a differential value having a high frequency of occurrence is associated with a code having a small number of bits, and a code corresponding to the computed differential value is read. In step S13, it is determined whether or not all normalization coefficients B have been encoded. If it is determined that all normalization coefficients B have been encoded, in step S14, the code read in step S12 is output. The present invention is applicable to an audio recorder.

Abstract: The inventive mechanism uses several sub-systems to generate outputs from the stereo input signal. A first sub-system synthesizes the phantom center output, which places the monaural center image between the left and right speakers in front of the listener. A second sub-system synthesizes the virtual surround (or rear) output signals, which places the sound images to the sides of the listener. A third sub-system synthesizes the left and right stereo outputs, and expands the locations of the left and right sound images.

Abstract: A stereo sound circuit device for providing three-dimensional surrounding effect has a difference circuit for acquiring difference signal of a left channel sound and a right channel sound. The stereo sound circuit device further provides a filter circuit consisting of a first and a second filtering operation units for applying a first and second filtering functions to the difference signal output from the difference circuit for generating a three-dimensional surrounding sound signal. The first filtering function implements a band pass filter with a central frequency of about 7.23 KHz. The second filtering function implements a band pass filter with a central frequency of about 318 Hz. A signal level adjust circuit is provided for adjusting the signal level of the sound signal output from the filter circuit. A mixer is provided for combining the adjusted three-dimensional surrounding sound with the original left channel sound and right channel sound.