Abstract: A stereophonic processing system provided with spatial expansion of the stereophonic sound so that a pair of spaced-apart loudspeakers will acoustically appear to be spaced apart further then they actually are. The spatial expansion circuit includes tonal compensation.
Abstract: Some workers wear headsets to protect their hearing from loud persistent noises, such as airplane engines and construction equipment. These headsets are generally passive or active, with the active ones including ear speakers and automatic noise-reduction (ANR) circuitry to cancel or suppress certain types of loud persistent noises. One problem with active headsets the concerns the difficulty of salvaging headsets that fail performance testing. Accordingly, the inventor devised a unique a unique cup-in-cup structure for the earcups of active headsets as well as related assembly and testing methods. The unique structure not only allows for pretesting of the ANR circuitry prior to assembly, but also enhances performance of the resulting headsets.
Abstract: An acoustic signal reproduction device includes a reproduction speaker and a signal processing section for generating an acoustic signal for causing a listener to recognize a reference speaker. The signal processing section includes a compensation data input section for receiving compensation data from the outside of the acoustic signal reproduction device and a calculation section for calculating the acoustic signal based on the audio signal and the compensation data and outputting the acoustic signal to the reproduction speaker. The compensation data has a value H/C, where H is a transfer function from the reference speaker to a control point located in the vicinity of an ear of the listener, and C is a transfer function from the reproduction speaker to the control point located in the vicinity of the ear of the listener.
September 19, 2001
Date of Patent:
May 4, 2004
Matsushita Electric Industrial Co., Ltd.
Abstract: An apparatus, system, and method for providing an audio interface between multiple deskset phones and a single radio antenna unit (RAU). The audio interface includes a transmit path and a receive path. The transmit path carries transmit signals from the multiple deskset phones to the RAU. The receive path carries a receive signal from the RAU to the deskset phones. The transmit path and receive path are electrically isolated from each other.
Abstract: A remote speaker system (14) for an automotive vehicle (10) has a holder (22) fixedly mounted within the vehicle (10). A speaker assembly (19) having a speaker housing (20) coupled to holder (22) and a speaker (26) coupled within the housing (20). A receive circuit (48) is coupled within the housing (20) and is electrically coupled to the speaker (26). A control circuit has an audio receiver generating audio signals and a transmit circuit (44) transmitting a wireless communication signal in response to the audio signal. The receive circuit (48) receives the wireless communication signal and converts the wireless communication signal into an audio electrical signal. The speaker converts the audio electrical signal into an audible signal.
Abstract: Predetermined frequency components contained in a luminance signal are extracted by BPF (12a)-(12n), and amplified by limited amplifiers (16a)-(16n). As a result, an addition signal, including a luminance signal component amplified and suppressed and a noise component amplified, is obtained from an adder (20a). The luminance signal is also supplied through an HPF (14) and adjusted in level by an amplifier (18). Therefore, a noise component is obtained from a subtracter (20b) by subtracting an amplified signal from the addition signal by a subtracter (20b). A subtracter (20c) subtracts this noise component from the luminance signal supplied from a delay circuit (22a), thereby outputting a luminance signal reduced of noise through an output terminal (S2). Because the luminance signal is separated into a plurality of bands by the plurality of BPFs, there is no possibility of saturating in noise component by the limiters, thus fully removing noise.
Abstract: A control device for an audio processor has a plurality of sets of function select controls and a function control section on its control device. Each set of the plurality of sets of function select controls is coupled with a corresponding channel in a plurality of channels associated with the audio processor. The sets include select switches for a predetermined set of functions which are executable in the corresponding channels. The select switches have first and second states which are visually distinguishable by an operator. The plurality of sets of function select controls are arranged on the control panel in a row, and the select switches in the plurality of sets are arranged in respective single columns within the plurality of sets so that the select switches for a particular function across the plurality of channels form a single band on the control panel.
Abstract: A method for determining the make up of a subscriber loop via improved time-domain reflectometry techniques by analyzing the echo responses generated by transmittal of pulses onto the subscriber loop. In the method discontinuities along a loop are identified sequentially and in a step-by-step fashion by comparing the measured waveform to suitable waveforms generated on the basis of a hypothesized topology. Once the generated waveform that best matches the measured data has been found and a discontinuity identified, the waveform generated in correspondence of the loop topology identified so far is subtracted from the measured data to produce a compensated waveform, which, is more suitable for detection and location of the next echo.
Abstract: A digital audio decoder decodes or expands compressed data such as bit stream data, which are compressed based on the MPEG/Audio standard. Inverse quantization circuits perform inverse quantization on plural bit stream data, which are supplied thereto in connection with multiple channels respectively, thus producing inversely quantized data with respect to a prescribed number (e.g., thirty two) of sub-band samples respectively. The inversely quantized data are combined together among the multiple channels with respect to the prescribed number of the sub-band samples respectively. Then, a filter bank synthesizes together combined data corresponding to all of the sub-band samples, thus reproducing original digital audio signals. Multipliers are provided for use in gain control on the inversely quantized data with respect to the sub-band samples respectively.
Abstract: The present invention concerns an adaptive headset interface for amplification and adjustment of signals between a random headset and a random terminal. The adaptive headset interface comprises digital signal processors, and is further equipped with a modem block over which the adaptive headset can be programmed by a call to a computer over the telephone net, whereby instructions as well as test signals are transferred between the computer and the adaptive headset interface, so that an adjustment is achieved which takes into account the signal transfer in relation to a reference telephone line corresponding to a given standard, or an adjustment which corresponds to the levels which apply for the relevant terminal to which the adaptive headset interface is connected.
Abstract: Handsfree telephony continues to be an increasingly desirable feature of modern telecommunications, whether in a conference room or mobile setting. Fundamental to the user acceptability of these systems is the performance of algorithms for acoustic echo cancellation, the purpose of which are to prevent the far-end signal from being transmitted back to the far-end talker. Most speech coding algorithms are based on some variant of Linear Predictive Coding (LPC), which reduces the amount of bits sent across a channel. Instead of doing echo cancellation in the time domain, the invention involves operating an acoustic echo canceller on the LPC parameters at the receiver, before the decoding stage.
December 15, 1999
Date of Patent:
April 6, 2004
Nortel Networks Limited
Andre J. Van Schyndel, Jeff Lariviere, Rafik Goubran
Abstract: The scope of the present invention is a device for detecting the source of a voice, which device comprises microphone means (2; 2a, 2b, 2M) for receiving a voice signal and detecting means for detecting the voice from the received voice signal. The device comprises means (15, 17) for determining the direction of arrival of the received signal, means (17) for storing the assumed direction of arrival of the voice of a certain source and means (18) for comparing the direction of arrival of said received signal with said assumed direction of arrival. The device further comprises means (18) for indicating that the source of the voice is said certain source when the comparison proves that the direction of arrival of said received signal matches with said assumed direction of arrival within a certain tolerance.
Abstract: The invention relates to a method for synthesizing a virtual sound source in a system (40) which comprises at least a right and a left channel for transmitting a stereo signal and in which the channels are connected to a filter block (42) for expanding the sound image. In the method, the amplifications of the separated monophonic and stereophonic signal components are optimized according to the stereophony of the signal coming to the system. The method according to the invention can also be applied to producing early room reflections by means of a separate filter block (71). The invention also relates to a device for synthesizing a virtual sound source, which device comprises at least a first and a second channel for transmitting the signal, at least one amplifier and filter and means for estimating the stereophony of the signal, for determining the amplification coefficient of the filtered signal and for controlling the amplifier according to the calculated amplification coefficient.
Abstract: Switch 141 continuously switches between received signal 2 and a supplemental signal, which is obtained by processing the received signal 2 through filter 145 to use the output of switch 141 in place of received signal 2. Accordingly, adaptive filters 122 and 124 operate sometimes by using received signal 2 as the input signal and sometimes by using the supplemental signal as the input signal, so that it is possible to obtain adaptive filter coefficients by using twice the number of conditional equations as the case of using only received signal 2 as the input signal. Therefore, since the adaptive filter coefficients do not becomes indefinite, it is possible to converge the coefficients to the correct values. Further, since switching period between the original and the supplemental signals is controlled to be longer than the sampling period of the received signal, it is possible to suppress aliasing distortion of the received signal directly supplied to a speaker and to be maintain better sound quality.
Abstract: To facilitate direct conversion to digital form of an acoustic signal acting on the acoustic receptor of an acoustic receiver while satisfying requirements of dynamic range, noise and adequate quantization, the following is proposed: the acoustic receptor should be exposed to a counter-signal when the acoustic signal acts on it in such a way that the acoustic receptor is largely maintained in its rest state despite the action of the acoustic signal. The counter-signal is derived from the control variable of a control circuit which is a component of the acoustic receptor. The control variable contains the information on the acting acoustic signal. Any deviation of the receptor from its rest state immediately generates a digital “nought” or “one.
Abstract: An odd-order low-pass microfilter is disclosed for being interposed between a home telephone wiring network and a POTS, or voice band, device to separate voice-band signals from higher frequency signals, such as ADSL signals and home networking signals. The filter topology is substantially symmetric so that the filter is reversible in that either end of the filter may be directly coupled to the home telephone wiring network without impairing high frequency signal performance or the filter characteristic of the filter. In one embodiment, the filter is a three-pole filter with a single capacitor disposed between a pair of coupled inductors. Each of the coupled inductors advantageously has an interwinding capacitance over about 100 pF to improve the filter frequency response without increasing the cost of the filter. In another embodiment, the filter is a reversible three-pole filter with a single capacitor disposed between first and second pairs of uncoupled, or discrete, inductors.
Abstract: Immersive environments for teleconferencing, collaborative shared spaces and entertainment require spatial audio. Such environments may have non-ideal sound reproduction conditions (loudspeaker positioning, listener placement or listening room geometry) where wavefront-synthesis techniques, such as ambisonics, will not give listeners the correct audio spatialization. A method disclosed for generating a sound field from a spatialized original audio signal, wherein the original signal is configured to produce an optimal sound percept at one predetermined ideal location. A plurality of output signal components are generated, each for reproduction by one of an array of loudspeakers. Antiphase output components are attenuated such that their contribution to the spatial sound percept is reduced for locations other than the predetermined ideal location.
July 9, 1998
Date of Patent:
February 17, 2004
British Telecommunications public limited company
Abstract: A speaker box for a sport utility vehicle. The speaker box is pivotally mounted to the sports bar of the vehicle and can be selectively pivoted relative to the bar to a plurality of positions. In one position, the sound from the speakers of the speaker box is directed downwardly as in convention designs. However, unlike conventional designs, the speaker box and its sound can also be directed rearwardly of the vehicle for parties and other gatherings such as at campsites or on the beach. Further, the speaker box and its sound can be directed slightly forward toward the front of the vehicle if desired. In doing so, a saddle or cutout is provided in the middle of the speaker box so as not to unduly inhibit the driver's vision through the rear view mirror.
Abstract: An intra-vehicle communication system includes a surface configured to be mounted to a vehicle interior proximate a first occupant wherein the surface is further configured to display an image of a second occupant of the vehicle. The system further includes an intercom including a microphone supported in close proximity with the surface and a speaker operably coupled to the microphone. In one embodiment, the intercom is further configured to at least partially mute existing audio output from a stereo system.
Abstract: A powered or active sub-woofer speaker system includes a power supply circuit, an audio amplifier circuit and a sub-woofer audio driver. The power supply circuit includes a toroidal power transformer to supply power to the audio amplifier circuit. The amplifier circuit includes a class-D audio amplifier adapted to amplify signals received from an audio source and to couple them to the sub-woofer audio driver. Each of the power supply circuit, audio amplifier circuit and sub-woofer audio driver has a low profile to permit the sub-woofer speaker system to be mounted in a closely confined space such as a house wall cavity.