Abstract: Improvements of a method and apparatus for insertion of a pseudo noise into a silent period between an audible period and a next audible period in digital audio communication between a transmitter 10 and a receiver 11. In the transmitter 10, level measuring circuit 1 measures the level of an incoming audio signal, and comparator 2 compares the measured level value with a predetermined reference value and outputs a binary signal representing an audio period or a silent period. First selector 4 selects, in an audio period, the input audio signal 21 but selects, in a silent period, the input audio signal 21 in a lower level, and outputs the selected signal as transmission data 22. First delay circuit 3 delays the output of comparator 2 by one word interval and outputs the delayed signal as a data enable signal 23 .

Abstract: An amplifier comprises a variable gain analog amplifier for amplifying an input analog signal to generate an intermediate analog signal, the gain of the analog amplifier being switchable between two or more discrete gain values. An analog to digital converter converts the intermediate analog signal into a corresponding intermediate digital signal and a variable gain digital amplifier amplifies the intermediate digital signal. A gain control circuit, responsive to a required gain, selects one of the discrete gain values of the analog amplifier and a gain value of the digital amplifier so that the sum of the selected gain values is substantially equal to the required gain; and whether the required gain is increasing or decreasing is detected.

Abstract: A bass audio circuit includes a first filter network which receives left and right channel input signals and produces two filtered signals by attenuating frequencies in the input signals above a first cutoff frequency. The two filtered signals are summed to create a first monophonic signal. A second filter network produces attenuated frequencies in either the input signals or in the first monophonic signal that are above a second cutoff frequency which is lower than the first cutoff frequency. The signals from the second filter network are combined and passed through a phase inverter to derive a second monophonic signal. An output stage individually amplifies the two monophonic signals which then either are applied to a dual voice coil woofer, or are electrically summed and applied to a single voice coil woofer. Alternatively, the two monophonic signals are first summed and applied to a single channel amplifier which drives a single voice coil woofer.

Abstract: A audio enhancement system and method of use with a sound system for producing primary sound from at least one main loudspeaker located at a main position. The audio enhancement system comprises at least one wireless transmitter, time delay circuitry, and plural augmented sound producing subsystems. Each sound subsystem is a portable unit arranged to be carried by a person located remote from the main loudspeaker and includes a wireless receiver and an associated transduce device, e.g., a pair of stereo headphones. The transmitter broadcasts an electrical signal which is representative of the electrical input signal provided to the main loudspeaker. The broadcast signal is receiver by the receiver and is demodulated and amplified to drive the transducer so that it produces augmented sound substantially in synchronism with the sound arriving from the main loudspeaker.

Abstract: A headphone includes an acoustic pipe, a loudspeaker unit, a microphone unit and a feedback circuit. The acoustic pipe has an inner diameter substantially equal to that of an external auditory canal. The acoustic pipe has a mounting portion provided at an end thereof for being mounted on the outer ear and has an acoustically non-reflective end at the other end thereof. The loudspeaker unit is provided on one side of the acoustic pipe with a sound emitting face thereof opposed to the inside of the acoustic pipe. The microphone unit is provided on the one side of the acoustic pipe with a sound collecting face thereof opposed to the inside of the acoustic pipe. The feedback circuit feeds back an output signal of the microphone unit to an input side of the loudspeaker unit. The feedback circuit includes an equalizing section. The output signal of the microphone unit is supplied to the loudspeaker unit by way of the equalizing section.

Abstract: A three-dimensional virtual audio display method is described which includes generating a set of transfer function parameters in response to a spatial location or direction signal. An audio signal is filtered in response to the set of transfer function parameters. The set of transfer function parameters are selected from or interpolatated among parameters derived by smoothing frequency components of a known transfer function over a bandwidth which is a non-constant function of frequency. The smoothing includes for each frequency component in at least part of the audio band of the display, applying a mean function to the amplitude of the frequency components within the bandwidth containing the frequency component, and noting the parameters of the resulting compressed transfer function.

Abstract: A three-way speaker comprises a high frequency transducer, a mid-frequency transducer, a low frequency transducer, and a hollow speaker cabinet defining a pyramidal mid-frequency transducer compartment and a low frequency transducer compartment isolated from the mid-frequency transducer compartment. The high frequency transducer is mounted in a first hole of the cabinet opening in the low frequency transducer compartment, the mid-frequency transducer is mounted in a second hole of the cabinet opening in the mid-frequency transducer compartment, and the low frequency transducer is mounted in a third hole of the cabinet opening in the low frequency transducer compartment. Therefore, the isolated mid-frequency transducer compartment and low frequency transducer compartment also isolates the mid-frequency transducer from a displacement of air produced by the low frequency transducer in the low frequency transducer compartment.

Abstract: A vibration control system includes a processor-based circuit which monitors and controls an analog vibration control circuit. The system includes a sensor to detect the undesirable vibrations and a synchronous pulse generator for determining the fundamental frequency of the vibrations emitted by the source. An actuator produces anti-noise to counter unwanted vibrations. The processor circuit tests the system, monitors system functions, and adjusts various parameters to provide optimal performance.

Abstract: An automatic sound controlling apparatus including a first adaptive filter means that forms a compensating signal for canceling a stationary sound, a signal producer means that produces a predetermined shape signal based on a timing signal with respect to a sound, a second adaptive filter means that forms a compensating signal for canceling a fluctuating sound by the predetermined shape signal, whereby in the case that a sound period changes sharply, an output signal of the second adaptive filter means is output in stead of the first adaptive filter means.

Abstract: A digital surround sound decoder. The decoder uses an architecture including two signal processing chips to achieve a program that can decode audio data at sufficiently high resolution. The decoder performs in real time and is compatible with standard surround sound formats. The decoder includes software that utilizes table lookups for critical functions in the decoding process. The processing flow of the decoder's program takes advantage of the multi-function capability within the specific processors used in the design while using a minimum number of program instructions. The program implements band pass filtering, sum-difference calculations, fast-attack slow-decay integration, summation and reciprocal processing, determination of fast and slow modes, look-up table indexing, adaptive matrix processing and various other functions to generate decoded surround sound signals from encoded left and right signal inputs.

Abstract: Method and apparatus for implementing surround sound in an audio system having a number of audio output devices, and audio output buffers associated with the audio output devices adapted to output audio data in order of receipt. A block of delay data is loaded into a buffer associated with a rear audio output device. Audio data is directed to the audio output devices, by way of the buffers. In the operation of the system, data stored in the buffers is transferred from the buffers to the associated audio output devices. The audio data will be delayed reaching the rear audio output device until the delay data has been output from the buffer resulting in a surround sound effect.

Abstract: A portable voice amplifier includes a casing providing a housing and including a flat base, a curved front wall extending from the base upwardly and over the base, when viewed from the side, to the top of the unit. A speaker is mounted in the lower portion of the curved front wall, and the casing includes a lower section partially formed by the lower portion of the curved front wall and having a progressively narrow depth preceding from the base to approximately the midsection of the casing. An upper section of the housing has generally uniform depth and width from the midsection to the top of the unit. The upper section is designed to be held conveniently by the hand of the user, with a microphone located on the rear of the upper housing section so that in normal use the entire unit extends below the face of the user and is conveniently and comfortably held with a single hand.

Abstract: In an active adaptive attenuation system having a main model, a second adaptive filter model is provided having an input from the output of the main model, and an output supplied to the input of the main model. A copy of the main model has an input from the input to the second model and an output supplied to a multiplier multiplying the error signal and the output of the copy and supplying the resultant product as a weight update signal to the second model.

Abstract: A composite loudspeaker apparatus including a cabinet which has a first air chamber and a second air chamber having a sound port, a drone cone or a resonance duct, adjacent to each other, wherein the first air chamber has a first loudspeaker driven by the audio signal of one of the left (L) and right (R) channels, and the second air chamber has a second loudspeaker driven by the low frequency component of the synthesized audio signal of the L and R channels mounted therein. The apparatus has another cabinet wherein a third loudspeaker driven by the audio signal of the other of the L and R channels is mounted therein. Another cabinet has a sound port, a drone cone or a resonance duct.

Abstract: A reverberation imparting apparatus comprises at least a data shift register, a coefficient shift register and an arithmetic convolution unit. The data shift register stores a string of sampling data each representative of an instantaneous value of an audio signal at each sampling period, while the coefficient shift register stores a string of coefficients which are created on the basis of waveforms of reflected sounds in response to an impulse sound artificially produced in a certain sound-field space such as a church. Normally, the data shift register successively stores new sampling data which are newly inputted thereto so that the contents of the data shift register is successively renewed. The arithmetic convolution unit performs an arithmetic convolution, using the string of coefficients on the string of sampling data.

Abstract: Audio apparatus demonstrating includes audio apparatus to be demonstrated. A display structure supports the audio apparatus and a digital computer controller. A CD-ROM is coupled to the audio apparatus and the digital computer controller with recorded sound signals for audible reproduction by the audio apparatus in response to selection signals furnished by the digital computer controller. A start demonstration switch is coupled to the digital computer controller. A proximity sensor provides a proximity signal to the digital computer controller when a person enters a predetermined zone around the apparatus to cause the digital computer controller to select a change in the record signal reproduced by the audio system from a first signal to a second signal which second signal encourages the person within the zone to actuated the start demonstration switch and initiate a demonstration of the audio apparatus including audibly reproducing sound signals recorded on the CD-ROM.

Abstract: The electronic stethoscope is designed to minimize the influence of the various types of noise while optimizing auscultation of the sounds of interest, and to enable a cardiologist to auscultate mechanical heart valves. It comprises a probe for sensing sounds of interest produced within a patient's body and for converting these sounds to an electric signal, and an audio amplifier and earphones for reproducing the sounds of interest in response to this electric signal.

Abstract: An active control system having an adaptive filter model in which the effect of transfer functions after the adaptive filter model output are eliminated from the adaption process. The adaptive control scheme inputs a reference signal to a copy of the adaptive filter model, and the copy of the adaptive filter model outputs a correction signal that drives an output transducer. The copy of the adaptive filter model does not receive an error input signal. Adaption occurs in the adaptive filter model. The model input to the adaptive filter model is a C-filtered reference signal. There are no transfer functions between the output of the adaptive filter model and the error input to the adaptive filter model. The error input to the adaptive filter model is a combination of an error signal from an error sensor, a C-filtered correction signal, and a model output signal from the adaptive filter model.

Abstract: A silencing apparatus for effectively reducing noises in the three dimension space which detects physical characteristics of the noise source by the laser doppler sensor unit during the noise source comes close to the telephone booth, and collects acoustic characteristics of the noise source by the sensor microphone array. The control portion generates an inverse-phased signal against the noise source so that the noises are decreased at the error sensor, and stores the noise source information ( physical characteristics and the acoustic characteristics) and the generated inverse-phased signal. When the same noise source come closely, the voice is generated from the speaker using the inverse-phased signal stored in the memory. The control portion generates an inverse-phased voice from the speaker when the collected noise by the sensor microphone is actually over a certain threshold level.

Abstract: A speaker driving circuit including a switch which switches the supplying of current to a speaker on and off by using a binary pulse-density-modulated (PDM) signal produced by .DELTA..SIGMA.-modulating a digital audio signal from a digital low-pass filter. For example, when the PDM signal is "1", the switch causes the output current of the direct-current power source to flow to the voice coil. On the other hand, when the PDM signal is "0", the switch stops the current to flow to the voice coil. This switch may be replaced with a switch which switches the direction of current flowing through the voice coil. The diaphragm of the speaker is driven by controlling the supplying of current in such a manner. With this arrangement, since a D/A converter for a digital audio signal is not required, there is no need to install an amplification circuit for an analog signal. Thus, the speaker driving circuit has a simplified structure but is capable of driving the speaker stably.