Abstract: Disclosed is a method of reducing clicking sounds in an audio data stream. The samples of the audio data are delayed by a predetermined amount in a sliding window containing a predetermined number of samples. The presence of a clicking sound in the delayed data within said sliding window is detected and the corresponding audio data replaced by substitute data derived from the audio data stream.

Abstract: A sound field measuring apparatus includes a microphone set having a first and second microphones arranged at a prescribed interval, which collects audio signals outputted from a first and second speakers, a measuring unit measuring distances between the first and second speakers, and the first and second microphones based on audio signals collected by the first and second microphones, and a position calculating unit calculating a position of the first and second microphones and a position of the second speaker when the first speaker is taken as a reference position based on the respective measured distances.

Abstract: In one embodiment, an apparatus comprises an adaptive filter, a timing recovery unit, and a reverse interpolation filter. The adaptive filter has adaptive filter coefficients that are adjusted based on a first error signal at a first sample rate and filters a first signal at the first sample rate to obtain a second signal at the first sample rate. The timing recovery unit interpolates the second signal at the first sample rate to obtain a third signal at a second sample rate; and estimates a partial response signal at the second sample rate corresponding to the third signal. The a reverse interpolation filter interpolates a second error signal at the first sample rate, which is a difference between the third signal and the partial response signal, to obtain the first error signal at the first sample rate for feeding back to the adaptive filter.

Abstract: A level ratio calculation circuit calculates average signal level data of signal level data corresponding to each sound collection beam signal, and calculates a level ratio between the average signal level data and each of the signal level data. Since a diffraction sound is substantially equal to all the signal level data, a diffraction sound component of the average signal level data also becomes substantially equal. On the other hand, a collection sound from a speaker is specific to the signal level data of the corresponding sound collection beam signal. Therefore, at the level ratio, the portion corresponding to the diffraction sound is flat and a data level becomes high locally in only the portion corresponding to the collection sound. By using this, the sound collection beam signal including the collection sound is detected.

Abstract: A system for adjusting a response of a loudspeaker system. The system includes a woofer installed in a housing. The woofer includes a woofer cone. A transmitter for transmitting an ultrasonic audio signal towards the woofer cone. A receiver for receiving the ultrasonic audio signal. A detection unit for determining the position of the woofer cone on the basis of a difference between the phase of the reflected ultrasonic audio signal with respect to the ultrasonic audio signal transmitted from the transmitter. A control unit for adjusting a spring constant of the membrane of the loudspeaker system on the basis of the determined position. A method for adjusting a response of a loudspeaker system, to a module to be used in the system. A computer program product including computer code for adjusting a response of the loudspeaker system.

Abstract: A method of modeling a room impulse response according to an embodiment of the invention includes receiving a sound pressure signal that is obtained by a microphone when an impulse-type sound source is excited and detecting a room impulse response; determining boundaries between a plurality of intervals of the room impulse response such that the room impulse response is divided into the plurality of intervals on a time domain; and modeling the room impulse response for each of the plurality of divided intervals using at least two different modeling schemes.

Abstract: A low-cost, real-time solution is presented for compensating memoryless non-linear distortion in an audio transducer. The playback audio system estimates signal amplitude and velocity, looks up a scale factor from a look-up table (LUT) for the defined pair (amplitude, velocity) (or computes the scale factor for a polynomial approximation to the LUT), and applies the scale factor to the signal amplitude. The scale factor is an estimate of the transducer's memoryless nonlinear distortion at a point in its phase plane given by (amplitude, velocity), which is found by applying a test signal having a known signal amplitude and velocity to the transducer, measuring a recorded signal amplitude and setting the scale factor equal to the ratio of the test signal amplitude to the recorded signal amplitude. Scaling can be used to either pre- or post-compensate the audio signal depending on the audio transducer.

Abstract: A portable terminal device able to suppress a feeling of strangeness caused by audio not desired by a user is provided. A mobile phone 2 has an audio output portion 33 outputting an audio signal, and can make a speaker 23 arranged in a cradle device 3 output audio in accordance with the audio signal therefrom by outputting the audio signal from the audio processing portion 33 to the cradle device 3 when it is connected to the connectable cradle device 3. The mobile phone 2 has an external speaker output control portion 43 for restricting the output of the audio signal to the speaker 23 when the type of the audio signal output from the audio processing portion 33 is a specific type.

Abstract: Sound can be listened to only in a listening area by maximizing a sound energy difference between a listening area and a non-listening area while maximizing sound radiation efficiency of each sound source. Accordingly, realistic sound can be provided to listeners without causing auditory disturbance to third parties, and maximal sound effects can be obtained with only minimal control.

Abstract: A loudspeaker control apparatus of the present invention includes: an operation reception section that receives a user operation; a mode selection section that selects between two operation modes, based on the user operation, either a normal mode in which operations of a plurality of loudspeakers and lights are controlled independently or an inspection mode in which each of the plurality of loudspeakers is inspected to see whether it is operating normally; and a control section that controls the operations of the plurality of loudspeakers and lights based on the operation mode, and the control section, in the inspection mode, sequentially causes the plurality of loudspeakers to output an inspection tone and sequentially causes lights which correspond to the loudspeakers from which the inspection tone is being outputted to be turned on or to blink.

Abstract: A variety of loudspeaker arrangements may have any of multiple adjacent magnetic circuits, single magnetic circuits, an improved loudspeaker voice coil assembly, multiple voice coil windings (124-127, 315-317), and commutated current. The radial direction of flux may alternate at adjacent poles and may have a controller that commands the current through each of the windings. The position of the moving components may be measured or inferred by the controller (213). An encoding track (318a) applied to the surface of the assembly may allow the assembly to function as part of a position transducer to permit appropriate action based on position. Calculated or sensed actual position may be used to determine the relative current in each of the windings and the controller may have compensation such as a motion control algorithm, thermal monitoring, and management of the driver.

Abstract: In public audio presentation it is desirable that the playback loudness is continuously adapted to the background noise loudness. Such adaptation of the playback volume requires a continuous measurement of the loudness somewhere within the sound exposure area. A dipole loudspeaker panel has a strong attenuation of the emitted sound in the panel plane. When placing a microphone capsule in that plane but outside the panel, the sound emitted from the panel will be recorded by the microphone with a significant sound pressure level attenuation. Such specific microphone arrangement can capture the background noise without being unduly disturbed by the direct sound emitted from the loudspeakers.

Abstract: An improved speakerphone for a cellular telephone, portable telephone handset, or the like. In one embodiment, a receiver provides an audio signal, and a first phase-shifter phase-shifts the audio signal by a first phase-shift amount. A second phase-shifter phase-shifts the audio signal by a second phase-shift amount and drives a loudspeaker. A processor sets the first phase-shift amount to each one of a plurality of phase-shift amounts and determines a corresponding average-to-peak ratio value of the first phase-shifted audio signal. The processor then selects one of the plurality of phase-shift amounts having a corresponding average-to-peak ratio value that meets at least one criteria (e.g., the largest one of the average-to-peak ratio values), and then sets the second phase-shift amount to be the same as the selected phase-shift amount. This enhances the perceived loudness of sound from loudspeaker.

Abstract: A headset is able to be coupled via a cable to an intercom system, is able to be wirelessly coupled to a wireless device via a wireless transceiver of the headset, and is able to be connected to a wired device via another cable. A controller of the headset separately monitors the microphone conductors and audio conductors by which the headset may be coupled to the intercom system to detect whether or not one or both of a communications microphone and an acoustic driver of the headset are coupled to the intercom system, and monitors the operating state of the wireless transceiver to detect whether or not the wireless transceiver is inactive, on standby or in use; and selectively couples a system ground conductor to one of the microphone conductors, and selectively provides a local microphone bias voltage in response to what is observed through such monitoring.

Abstract: An audio power management system manages operation of audio devices in an audio system. The audio power management system includes a parameter computer, a threshold comparator and a limiter. Audio signals generated with the audio system may be provided to the audio power management system. Based on a measured actual parameter of the audio signal, such as a real-time actual voltage and/or a real-time actual current, the parameter computer can derive estimated operational characteristics of audio devices, such as a loudspeaker included in the audio system. The threshold comparator may use the estimated operational characteristics to develop a threshold and manage operation of one of more devices in the audio system by monitoring the measured actual parameter, and selectively directing the limiter to adjust the audio signal, or another device in the audio system to protect or optimize performance.

Abstract: There is provided a multichannel audio amplification apparatus including a speaker detection circuit for detecting whether or not a speaker is connected to an arbitrary speaker terminal of speaker terminal groups TA and TB, and an operating circuit and a display circuit connected to a control circuit. Even in a case where one of two speaker groups of A and B types connected to the multichannel audio amplification apparatus cannot be connected with a surround speaker for reproducing only a surround sound, a listener is prevented, by indication, from carrying out on the operating circuit an erroneous operation or setting which otherwise might be liable to occur both in a case where the B-type speaker group including no surround speaker is connected to the multichannel audio amplification apparatus and in a case where the B-type speaker group is not connected thereto.

Abstract: The present invention relates to a method of performing measurements by means of an audio system comprising passive loudspeakers, whereby said measurements loudspeakers for producing sound and at least one of said loudspeakers for measuring said sound. The present invention further relates to an audio system comprising N passive loudspeakers, wherein said audio system further comprises an output stage where each output acts as a combined output channel and a measurement input.

Abstract: A system quantifies listener envelopment in a loudspeakers-room environment. The system includes a binaural detector that receives frequency modulated audible noise signals from multiple loudspeakers. The binaural detector generates detected signals that are analyzed to determine an objective listener envelopment. The envelopment is based on binaural activity of one or more sub-bands of the detected signal.

Abstract: A device for testing the operability of an audio speaker includes a housing, a unidirectional microphone and a display. The underside of the housing comprises a recessed area. The unidirectional microphone is positioned and oriented in the recessed area such that the unidirectional microphone is recessed from the underside of the housing. The display is positioned on the housing such that the display is visible when the testing device is placed proximate an audio speaker with the underside of the housing facing the audio speaker. The display may include at least one visual indicator operatively coupled to the unidirectional microphone such that, when the unidirectional microphone is receiving an acoustic signal, the at least one visual indicator is illuminated.

Abstract: A public address system includes at least one monitoring device, a transmission medium and at least one level monitoring device. The monitoring device is connected to the transmission medium and powered by an AC voltage with a supply frequency present on the transmission medium. The monitoring device receives or transmits or transmits and receives a communication signal utilizing a carrier frequency upon the transmission medium. The carrier frequency is different from the supply frequency and the level monitoring device is adapted to monitor the level of the AC voltage.

Abstract: Aspects for monitoring audible tones indicative of operational status of each planar in a multiple planar chassis are described. Included in the aspects is the monitoring of a speaker channel of each planar of a plurality of planars in a common chassis for state changes of beep tones. An operational status of a specific planar emitting the beep tones is identified based on the state changes.

Abstract: A microphone-capsule includes a vibrator implemented by a conductor having a flat vibration surface, a dielectric-polarization plate that is defined by a flat first principal surface facing the vibration surface of the vibrating plate and a second principal surface facing parallel to the first principal surface and in which polarization directions are aligned, a back electrode joined to the second principal surface of the dielectric-polarization plate, and an induced-charge measuring unit configured to measure the charges induced between the vibrating plate and the back electrode, in association with the displacement of the vibration surface. The induced-charge measuring unit includes an amplifier connected to the back electrode and an output circuit connected to the amplifier.

Abstract: A sound device includes an input terminal for electronic signal input; a line out terminal adapted to output electronic signals input on the input terminal to an external area; a delay arrangement adapted to delay electronic signals input on the input terminal for a certain time; a main output arrangement adapted to output electronic signals delayed by the delay arrangement. The delay arrangement delays signals in order to accommodate a time difference between (1) electronic signals output from an external sound device connected with the line output terminal and (2) electronic signals output from the main output arrangement. The time difference between sound that is output from the sound device and the one that is output from an external sound device that is connected with line connection terminal can be eliminated. Thus, it can prevent sound quality decrease due to time difference from happening.

Abstract: In a voice intelligibility enhancement system that controls a gain of a voice signal based on noise power and voice power of the voice signal generated by a voice signal generation unit, it is detected whether the voice power is equal to or greater than a predetermined level, noise power output when the voice power is less than the predetermined level is measured and stored, noise power to be output when the voice power exceeds the predetermined level is estimated to be the stored noise power, and gain of a voice signal is controlled on the basis of the voice power and the estimated noise power.

Abstract: A method for controlling a maximum signal level output to headphones of a wireless device is provided. The method includes: determining an impedance of the headphones; determining a carrier specific maximum signal level for headphones having the impedance; and, adjusting an audio amplifier of the wireless device coupled to the headphones to restrict the maximum signal level output to the headphones to the carrier specific maximum signal level.

Abstract: The present invention provides a powered speaker, such as a subwoofer that may be turned on and off remotely by energizing the auxiliary AC power supply provided with an audio device or other source unit. The main power of the powered subwoofer is provided via a wall outlet or suitable power supply that may adequately supply the necessary power.

Abstract: The present invention provides a module for a powered speaker, such as a subwoofer that may be actuated to turn the speaker on and off remotely by energizing the auxiliary AC power supply provided with an audio device or other source unit. The main power of the module is provided via a wall outlet or suitable power supply that may adequately supply the necessary power.

Abstract: The present invention relates to a method and corresponding system for predicting the perceived spatial quality of sound processing and reproducing equipment. According to the invention a device to be tested, a so-called device under test (DUT), is subjected to one or more test signals and the response of the device under test is provided to one or more means for deriving metrics, i.e. a higher-level representation of the raw data obtained from the device under test. The derived one or more metrics is/are provided to suitable predictor means that “translates” the objective measure provided by the one or more metrics to a predicted perceived spatial quality. To this end said predictor means is calibrated using listening tests carried out on real listeners. By means of the invention there is thus provided an “instrument” that can replace expensive and time consuming listening tests for instance during development of various audio processing or reproduction systems or methods.

Abstract: A source width estimation system may estimate an auditory source width of a sound. The system detects a first frequency modulated noise signal and a second frequency modulated noise signal. The detected noise signals may be processed to obtain an average value of a lateral width of a binaural activity over time and a plurality of sub-bands. The auditory source width of the noise signals may be determined from the average value of the lateral width of the binaural activity.

Abstract: A method of adjusting a loudness of an audio signal may include receiving an electronic audio signal and using one or more processors to process at least one channel of the audio signal to determine a loudness of a portion of the audio signal. This processing may include processing the channel with a plurality of approximation filters that can approximate a plurality of auditory filters that further approximate a human hearing system. In addition, the method may include computing at least one gain based at least in part on the determined loudness to cause a loudness of the audio signal to remain substantially constant for a period of time. Moreover, the method may include applying the gain to the electronic audio signal.

Abstract: A method of limiting the maximum permitted sound volume in an earphone, a headset or the like, includes the step that the size of the driving signal for driving the earphone is sensed. The driving signal has a voltage that is less than approx. 0.7 V. If the driving signal exceeds a certain level, one part of the driving signal is shunted past the earphone and another part of the driving signal is diverted. Either the voltage of the diverted part is increased and controls the shunt device by which the first part of the driving signal is shunted past the earphone, or it is supplied to a unit with a detector device and the shunt device. The unit is sensitive to work at voltages below approx. 0.7 V. An earphone or the like for the above-mentioned method comprises a shunt device and a detector device whose current source is the signal source of the earphone.

Abstract: A mobile communication device includes a loudspeaker for reproducing speech from a speech signal, a vibrator, and a measuring unit for measuring background noise in relation to the reproduced speech. The communication device further includes a vibrator processing unit for generating a control signal dependent on the background noise for controlling operation of the vibrator during speech reproduction dependent on a level of the background noise.

Abstract: An electroacoustic transducer includes a housing and a moving portion adapted to move relative to the housing. The moving portion has a first surface and includes a first electrode. A second electrode on a first surface of the housing is located proximate to a first region of the first electrode, and a third electrode on the first surface of the housing is located proximate to a second region of the first electrode. A capacitance between the first and second electrodes and a capacitance between the first and third electrodes vary similarly with displacement of the moving portion relative to the housing. An impedance buffer is coupled to the second electrode and the third electrode.

Abstract: A portable communication device is disclosed. The portable communication device comprises a speaker adapted to be held to an ear of a user for conveying sound to the user, at least one sensor for sensing sound emanating from said sound conveyed to the user, and a control unit. The control unit is adapted to estimate, based on an electrical input signal supplied to an input port of the speaker and an electrical output signal received from an output port of the at least one sensor, a transfer characteristic from the input port of the speaker to the output port of the sensor. Furthermore, the control unit is adapted to estimate, based on the estimated transfer characteristic, a degree of sound leakage from the user's ear. A corresponding method is also disclosed.

Abstract: A system and method for minimizing the complex phase interaction between non-coincident subwoofer and satellite speakers for improved magnitude response control in a cross-over region. An all-pass filter is cascaded with bass-management filters in at least one filter channel, a[1d preferably all-pass filters are cascaded in each satellite speaker channel. Pole angles and magnitudes for the all-pass filters are recursively calculated to minimize phase incoherence. A step of selecting an optimal cross-over frequency may be performed in conjunction with the all-pass filtering, and is preferably used to select an optimal cross-over frequency prior to determining all-pass filter coefficients.

Abstract: A spatial sound energy (SSE) distribution control apparatus calculates filter coefficients for controlling distribution of the sound energy of an input signal, in consideration of a sound energy ratio between a reduction region for reducing transmission of a sound energy emitted through an array speaker and a concentration region for concentrating transmission of the sound energy and also in consideration of a sound energy efficiency of the concentration region. Also, the SSE distribution control apparatus determines an array size of a speaker in a case where the sound energy ratio is maximized, according to frequency variation of the input signal.

Abstract: Embodiments of the invention can relate to methods and devices for determining a room acoustic impulse response in a time domain. In one embodiment, an acoustic input signal from an acoustic signal source is emitted into an acoustic room, an acoustic output signal is detected by an acoustic measuring device in the room and fed from the acoustic measuring device to an evaluating device and, via the evaluating device, from a reference signal corresponding to the acoustic input signal, and the acoustic output signal, if necessary after prior processing of the acoustic output signal, a room acoustic impulse response in the time domain of the acoustic room is calculated in realtime and prepared for output in that, temporally in parallel and continuously, the acoustic input signal is emitted, the acoustic output signal is detected and, via the evaluating device, the room acoustic impulse response in the time domain is determined.

Abstract: A system and method is disclosed for harmonizing calibration of audio between a plurality of networked conference rooms to enable each networked conference room to have substantially similar audio characteristics by adjusting speaker gain output of an audio signal sent from a calibration location on a network and tuning microphone response received at a calibration location to a calibrated audio source.

Abstract: A method for spotting an interaction in which a target speaker associated with a current index or current interaction speaks, the method comprising: receiving an interaction and an index associated with the interaction, the index associated with additional data; receiving the current interaction or current index associated with the target speaker; obtaining current data associated with the current interaction or current index; filtering the index using the additional data, in accordance with the current data associated with the current interaction or current index, and obtaining a matching index; and comparing the current index or a representation of the current interaction with the matching index to obtain a target speaker index.

Abstract: In one embodiment the present invention includes a method of wireless communication. The method comprises receiving a sonic signal and determining a sequence of sonic tones from a received sonic signal. The receiving includes receiving the sonic signal at an electronic device using a microphone. The sonic signal includes a sequence of sonic tones. The receiving results in the received sonic signal. The sequence of sonic tones contains predefined timing. The timing includes the duration of each sonic tone and a set of intervals between successive sonic tones.

Abstract: A method and system for determining and compensating for a non-linearity in a hands-free acoustic telecommunication device is disclosed. The method determines a back electromotive force signal induced in a loudspeaker from at least one of a coil voltage, a current signal and estimates of coil resistance and inductance, estimating at least one of a cone position and a cone velocity from the BEMF integrated with respect to time and determining an estimate of an echo value from a series connection of an estimated inverse of a force factor function primitive and the estimated acoustic impulse response; and outputting the estimated echo value.

Abstract: The present invention provides a sound producing apparatus which suppresses an increase in electric power consumption and cost, and an increase in a number of wiring when a plurality of speaker apparatuses is configured. A speaker driving apparatus supplies electric power to a speaker apparatus, acquires information about an electroacoustic characteristic and maximum input electric power of each speaker apparatus from the speaker apparatus, and executes characteristic correction and output electric power restriction according to the information.

Abstract: A signal processing apparatus includes a plurality of equalizers configured to input an audio signal of a corresponding channel among audio signals of a plurality of channels and configured to perform at least gain adjustment on the basis of a set parameter, each of the equalizers being provided in such a manner as to correspond to an audio signal of one of the plurality of channels; a plurality of output sections configured to output each audio signal for each of the plurality of channels, the audio signal being processed by the equalizer; a measurement section configured to measure frequency-amplitude characteristics of the audio signal output from the output section; and a computation section configured to perform a computation process for correcting frequency-amplitude characteristics of an audio signal of each channel on the basis of the measurement result by the measurement section.

Abstract: With an array of speakers including a center speaker provided with left and right ultrasonic electro-acoustic transducers and left and right speakers provided with respective ultrasonic electro-acoustic transducers, it is possible to identify the left and right speakers. One approach includes energizing the left transducer of the center speaker to emit an acoustic ping signal, utilizing the transducers of the left and right speakers to detect the ping signal, measuring lapse of time between emission of the ping signal by said the left transducer and detection of the ping signal by the transducers of the left and right speakers. Then, the right transducer of the center speaker is energized to emit an acoustic ping signal, the transducers of the left and right speakers are utilized to detect the ping signal, and lapse of time between emission of the ping signal by the right transducer and detection of the ping signal by the transducers of the left and right speakers is measured.

Abstract: Systems and processes for compensating for changes in a theatre sound system positioned in a theatre are described. A subsequent response of a loudspeaker to a test signal is captured and compared to a previously obtained signature response of the loudspeaker to the test signal. An audio signal can be processed based on the comparison to compensate for changes to loudspeaker performance, or otherwise.

Abstract: A voice processing apparatus includes a band dividing portion dividing a first voice signal generated by a first microphone and a second voice signal generated by a second microphone into predetermined frequency bands, a sound source segregating portion segregating an echo component of a voice emitted by a first sound source included in a voice emitted by a second sound source in each of the predetermined frequency bands based on the power of the first and second microphones, and a band synthesis portion synthesizing the first and second voice signals from which the echo component of the first sound source has been segregated by the sound source segregating portion into a voice signal including the voice emitted by the first sound source and a voice signal including the echo component of the first sound source.

Abstract: A sound system obtains a desired sound field from an array of sound sources arranged on a panel. The desired sound field allows a listener to perceive the sound as if the sound were coming from a live source and from a specified location. Setup of the sound system includes arranging a microphone array adjacent the array of sound sources to obtain a generated sound field. Arbitrary finite impulse response filters are then composed for each sound source within the array of sound sources. Iteration is applied to optimize filter coefficients such that the generated sound field resembles the desired sound field so that multi-channel equalization and wave field synthesis occur. After the filters are setup, the microphones may be removed.

Abstract: An audio power management system manages operation of audio devices in an audio system. The audio power management system includes a parameter computer, a threshold comparator and a limiter. Audio signals generated with the audio system may be provided to the audio power management system. Based on a measured actual parameter of the audio signal, such as a real-time actual voltage and/or a real-time actual current, the parameter computer can derive estimated operational characteristics of audio devices, such as a loudspeaker included in the audio system. The threshold comparator may use the estimated operational characteristics to develop a threshold and manage operation of one of more devices in the audio system by monitoring the measured actual parameter, and selectively directing the limiter to adjust the audio signal, or another device in the audio system to protect or optimize performance.

Abstract: A sound monitoring system and method. The system can include a plurality of sound pressure level meters, a plurality of sound level indicators and a server connected by a network. The sound pressure level meters measuring a sound level at their location, and the sound level indicators providing a visual indication of the sound level measured by at least one of the sound pressure level meters. The system devices can be powered, as well as monitored and controlled remotely over the network. A user interface enables constructing and monitoring multiple zones and groups in a monitored area, as well as reviewing real-time and historical sound data. The system can also control lighting in the monitored area, and use lighting as a visual indicator of noise level.

Abstract: A method for controlling a maximum signal level output to headphones of a wireless device is provided. The method includes: determining an impedance of the headphones; determining a carrier specific maximum signal level for headphones having the impedance; and, adjusting an audio amplifier of the wireless device coupled to the headphones to restrict the maximum signal level output to the headphones to the carrier specific maximum signal level.