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 method for showing an array microphone effect includes the steps of obtaining an original acoustic signal from array microphones, and visualizing the original acoustic signal to obtain a figure. The original acoustic signal includes a crystal voice, out-beam noises, background noise, and/or an echo. The figure includes a plurality of graphic components representing the crystal voice, out-beam noise, background noise, and/or echo, respectively.

Abstract: An audio processor device and method is disclosed which measures and provides information relating to the audio level being applied to the ear of a user. The processor device uses a preset or calibrated sensitivity of the applied earphones in combination with an analysis of the audio stream to provide sound-pressure-level or time-weighted exposure information to the user or limit the output when preset levels have been achieved. Also disclosed is the use of microphones, internal or external, to combine an additional audio stream, typically the ambient environment, into the main audio channel.

Abstract: An example game apparatus generates and outputs a predetermined test image to a television. A terminal device has its image pickup section acquire a pickup image of a screen of the television, and transmits the pickup image acquired by the image pickup section to the game apparatus. The game apparatus determines whether or not the pickup image includes the test image. When the pickup image is determined to include the test image, an image delay time is calculated on the basis of the time of the determination, the time of the output of the test image by the game apparatus, and a processing time between the acquisition of the pickup image and the determination. The game apparatus uses the image delay time to achieve synchronization between the terminal device and the television and also between image display and sound output of the television.

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, selectively provides a local sidetone, and/or selectively provides a local microphone bias voltage in response to what is observed through such monitoring.

Abstract: A speech communications system for a vehicle includes a microphone system provided in the vehicle interior in order to detect audio information. An interaction manager provides grammar information to a speech recognizer. The speech recognizer provides speech recognition results to the interaction manager. An acoustic echo canceller eliminates portions of the audio information detected by the microphone system. A sound localizer determines a sound source location in the vehicle interior. A method of operating a speech communications system in a vehicle is also provided. An interruptible text-to-speech operation provides a speech output to a user. Voice information is requested from the user for a maximum number of times if insufficient voice information or no voice information is provided in response to the speech output provided by the interruptible text-to-speech operation. The dialog context of an unfinished speech interaction is saved.

Abstract: The size of measuring boxes for hearing apparatuses and in particular for hearing devices is to be reduced, with the efficiency of the measuring box in respect of attenuating interference noises being maintained or improved. A measuring box is thus proposed, which comprises an interference signal recording facility for recording an interference signal. Furthermore, provision is made in the measuring box for a signal generating facility to generate a compensation signal which is phase-opposed to the recorded interference signal, so that the interference signal can be compensated for by the compensation signal. The interference noise attenuation is thus achieved here by an electronic active part, so that the quality demands on the measurement space can be reduced and its size thereby decreased as well.

Abstract: Practical speaker connection is identified using a device having a sound channel of a 5.1 channel or 7.1 channel, and a device is provided that can easily reproduce the optimum multiple channels. Actual speaker arrangement can be identified by, for example, measuring the impedance of a terminal at the side of an audio amplifier. If incorrect connection is found, a warning is issued. This information is transmitted to a signal source with an EDID and a signal with the optimum a number of sound channel is sent. The EDID is also used for the connection with a display unit and the speaker connection with which the display unit is provided uniquely. For example, a sound through the 7.1 channel is easily reproduced using the speaker of the display unit in the channel of the front speaker.

Abstract: A system for monitoring sound pressure levels at the ear includes an ambient sound microphone (ASM) for receiving ambient sounds and an ear canal receiver (ECR) for producing audio signals as a function of ambient sound received at the ambient sound microphone and a sound signal received from an associated personal audio device. A logic circuit is operatively associated with the ASM and ECR and calculates a total SPL_Dose experienced by the ear for periods ?t. The logic circuit may select an action parameter in response to the Total SPL_Dose.

Abstract: In one of many possible embodiments, a method includes providing an audio output signal to an output device for broadcast to a user, receiving audio input, the audio input including user voice input provided by the user and audio content broadcast by the output device in response to receiving the audio output signal, applying at least one predetermined calibration setting, and filtering the audio input based on the audio output signal and the predetermined calibration setting. In some examples, the calibration setting may be determined in advance by providing a calibration audio output signal to the output device for broadcast, receiving calibration audio input, the calibration audio input including calibration audio content broadcast by the output device in response to receiving the calibration audio output signal, and determining the calibration setting based on at least one difference between the calibration audio output signal and the calibration audio input.

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: An audio apparatus includes a sound generation unit configured to generate a measuring signal, a sound collection unit configured to obtain a sound collection signal, a detection unit configured to detect a level difference between the sound collection signal and background noise, a characteristic calculation unit configured to extract each signal period of the measuring signal from the sound collection signal, a feature amount calculation unit configured to calculate a feature amount representative of sudden noise for each period based on each of the period signals, and a determination unit configured to compare feature amounts of the periods and to remove, from periods to be used for averaging the period signals, a period or periods whose feature amounts are not within a range between a threshold value and a minimum value of the feature amounts as a reference, wherein the threshold value is determined according to the detected level difference.

Abstract: An audio testing system includes an acquisition unit, an input unit, a first analysis unit, a second analysis unit, and an output unit. The acquisition unit obtains an audio signal played back by the audio player. The input unit generates a first input signal and a second input signal. The first analysis unit analyzes the acquired audio signal with a first analysis method in response to the first input signal to obtain performance parameters of the audio player. The second analysis unit analyzes the acquired audio signal with a second analysis method in response to the second input signal to obtain another set of performance parameters of the audio player. The first analysis unit is more accurate than the second analysis unit, and efficiency of the second analysis unit exceeds that of the first analysis unit.

Abstract: An electronic device and method for testing an audio module of a network device including establishing a first connection between an audio output interface of the electronic device and a microphone interface of the network device. First audio data of a test file is transmitted from the audio output interface to the microphone interface through the first connection. An audio card of the network device records the first audio data output by the microphone interface to generate second audio data, and transmit the second audio data to the electronic device through a network. The second audio data is compared with the first audio data in response to receiving the second audio data by the electronic device. The audio module is determined to be normal if the second audio data matches the first audio data.

Abstract: A system locates a speaker in a room containing a loudspeaker and a microphone array. The loudspeaker transmits a sound that is partly reflected by a speaker. The microphone array detects the reflected sound and converts the sound into a microphone signal. A processor determines the speaker's direction relative to the microphone array, the speaker's distance from the microphone array, or both, based on the characteristics of the microphone signals.

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 disclosed example method to determine a location of an audience member involves generating a correlation analysis result based on correlating first audio samples from a stationary audio detector with second audio samples from a portable audio detector carried by the audience member and determining via a neural network the location of the audience member based on the correlation analysis result.

Abstract: An apparatus, method, and system for analyzing the response of a device under test using a signal generated by an audio card comprises a housing. The apparatus comprises a first connector coupled to the housing and configured to couple electrically with an output of the audio card and to receive a first signal from the audio card. The apparatus also comprises a second connector being electrically coupled with the first connector and configured to interface with the device under test in order to transmit the first signal to a device under test. The apparatus also comprises a third connector configured to receive a second signal from the device under test. The apparatus also comprises a fourth connector being electrically coupled with the third connector and configured to transmit the second signal to an input of the audio card.

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: Methods of operating an audio device are provided. A method includes measuring sound pressure levels (SPLECM) for acoustic energy received by an ear canal microphone (ECM) during a time increment ?t; and calculating a SPL_Dose?t during the time increment ?t using SPLECM.

Abstract: Methods of operating an audio device are provided. A method includes calculating estimated sound pressure levels (SPLs) for drive signals directed to an ear canal receiver (ECR) during a time increment ?t; calculating an estimated SPL_Dose during the time increment ?t using the estimated sound pressure levels; and calculating a total SPL_Dose at a time t of the audio device using the estimated SPL_Dose.

Abstract: Methods and devices for sound detection and audio control are provided. A listening device (100) can include a receiver (102) and a sound director for directing a sound produced by the receiver into an ear of the user, a microphone (104) and a mount for mounting the microphone so as to receive the sound in an environment, a detector for detecting an auditory signal in the sound received by the microphone, and an alerting device for alerting the user to the presence of the auditory signal. The user's personal safety is enhanced due to the user being alerted to the presence of the auditory signal, which otherwise may be unnoticed by the user due to a loud sound level created at the ear of the user by the receiver.

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 checking availability of audio interface between a display apparatus and an AV apparatus in an AV system, and an AV apparatus using the same are provided. The checking includes transmitting a command to check an audio interface to a second AV apparatus where an audio from a first AV apparatus is output, and receiving a result of checking the audio interface in response to the command from the second AV apparatus. Therefore, a user can recognize whether or not an audio interface has a problem in an AV system.

Abstract: The present disclosure provides a method of determining microphone polarity in a headset. By being able to detect the polarity of the microphone within the headset, a portable electronic device may be compatible with any headset and is not limited to use with a specific headset. By applying a detection signal over one of lines within a headset cable, a return signal may be sensed on one of the other lines to determine which of the lines is the ground line and which line is the microphone line.

Abstract: Speakers are identified based on sound origination detection through use of infrared detection of satellite microphones, estimation of distance between satellite microphones and base unit utilizing captured audio, and/or estimation of satellite microphone orientation utilizing captured audio. Multiple sound source localization results are combined to enhance sound source localization and/or active speaker detection accuracy.

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: One embodiment of the present invention sets forth a method for managing a power state of an audio device resident in a graphics processing unit. The method includes the steps of directing audio data originated from a client application via an audio path in an audio driver stack to the audio device, determining whether an active stream of audio data along the audio path is present in response to a notification of an attempt to shut down the graphics processing unit, and requesting a plug and play manager to disable the audio device, if no active stream of audio data is present along the audio path.

Abstract: Systems, methods, apparatus, and machine-readable media for detecting head movement based on recorded sound signals are described.

Abstract: A howling canceller applied to an acoustic system having a speaker and a microphone comprises: a filter insertion unit for inserting a notch filter at a frequency of an audio signal picked up by the microphone; a setting unit for setting the insertion time of the notch filter on the basis of the frequency at which the notch filter is inserted; and a releasing unit for, when the insertion time set by the setting unit has elapsed, releasing the notch filter, the insertion time of which has elapsed. The setting unit sets the insertion time of the notch filter to be shorter as the frequency at which the notch filter is inserted increases.

Abstract: A plural-channel audio signal (e.g., a stereo audio) is processed to modify a gain (e.g., a volume level or loudness) of an estimated dialogue signal (e.g., dialogue spoken by actors in a movie) relative to other signals (e.g., reflected or reverberated sound). In some aspects, a controller is used to control master volume and dialogue volume. In some aspects, one or more graphical objects and/or user interface elements are used to indicate volume levels and other information.

Abstract: The invention makes use of impulse responses of the performance venue to process a recording or other signal so as to emulate that recording having being recorded in the performance venue. In particular, by measuring or calculating the impulse responses of a performance venue such as an auditorium between an instrument location within the venue and one or more soundfield sampling locations, it then becomes possible to process a “dry” signal, being a signal which has little or no reverberation or other artifacts introduced by the location in which it is captured (such as, for example, a close microphone studio recording) with the impulse response or responses so as to then make the signal seem as if it was produced at the instrument location in the performance venue, and captured at the soundfield sampling location.

Abstract: Methods, systems, and computer programs for monitoring quality of audio delivered over a communications channel are presented. One method includes an operation for defining timestamps. The timestamps are associated with a measure of time while delivering audio to a client computer, where each timestamp includes a plurality of timestamp bits. Further, the method includes an operation for modulating an audio signal with pseudo noise (PN) codes when a timestamp bit has a first logical value, and modulating the audio signal with a negative of the PN codes when the timestamp bit has a second logical value. After transmitting the modulated audio signal to the client computer, the timestamp bits are extracted from a received modulated audio signal to obtain received timestamps. The quality of the audio is assessed based on the received timestamps, and the quality of the audio is stored in computer memory.

Abstract: In an audio output system including a TV, an AV amplifier, a DVD player, and the like, a simple control program decides an audio output mode setting, i.e., whether to output audio from a built-in speaker of the TV or output audio from an external speaker connected to the AV amplifier by linking operations of the respective devices. A control unit of a DVD player transmits a request signal concerning the audio output mode to a control unit of a TV. According to a request from the DVD player, the control unit of the TV instructs a control unit of an AV amplifier to be set to a mode for outputting audio from an external speaker connected to the AV amplifier or a mode for not outputting audio and sets a built-in speaker of the TV to a state opposite to the state of the external speaker.

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: The invention relates to simulation of sound fields in enclosures, for instance for application in listening tests, where test subjects assess the sound quality or other sound perception characteristics of the sound field. According to a specific embodiment, the system comprises a binaural synthesis portion which synthesizes sound for instance from a sound-reproduction equipment based on measured impulse responses of an actual room stored in a data base (31) and a binaural recording portion comprising a data base 32 for storing binaural recordings of other sound signals made in the room. Data from these databases are mixed (41) and reproduced by means of a headphone (39) provided with a head tracker (42) for tracking the movements of the listener's head. The invention furthermore comprises the use of cross-fading functions (36, 37) to enable the dynamic listening conditions, where the movements of the listener's head are taken into account during the simulation process.

Abstract: A device and a method for analyzing sound based on a biomorphic design are disclosed. The device comprises a plurality of amplification/filtering stages (S1, . . . , Sn) connected in a series configuration. Each amplification/filtering stage comprises at least one nonlinear amplification module (100a, . . . , 100c), preferably a Hopf amplifier, and at least one filter module (200a? . . . , 200c; 200a?, . . . 200c?) providing high-frequency attenuation.

Abstract: A device and method for providing augmented reality enhanced audio. The device is portable and may include a location processing module capable of receiving location information and determining a location of the device in an environment, an orientation processing module capable of determining an orientation of the device at the location, a processor capable of modifying audio content based on the location of the device in the environment and the orientation of the device at the location, and an audio output device capable of outputting the modified audio content. The orientation of the device includes an azimuth and an elevation of the device.

Abstract: According to an aspect of the invention, a portable terminal includes: a first audio input-output unit and a second audio input-output unit configured to receive audio when a mode is set as an input mode and output audio when the mode is set as an output mode; and a setting unit configured to set the respective modes of the first audio input-output unit and the second audio input-output unit.

Abstract: An audio electronic device includes a player, an output device, a detector, and an automatic turn-off module. The player is for reproducing audio data. The output device is for transforming the audio data to audible sound. The detector is for detecting positional shift of the output device. The automatic turn-off module is for turning off the player based on the positional shift. An automatic turn-off method is also disclosed.

Abstract: A sound characteristic parameter is obtained for an audio signal to be analyzed by analyzing the signal by means of a sound characteristic parameter analyzing section arranged upstream relative to a similarity determining section. A sound signal operation as retrieving key is read out from a retrieving key storage area of a retrieving key memory by a retrieving key audio signal reading section and analyzed by a retrieving key sound characteristic parameter analyzing section to obtain a retrieving key sound characteristic parameter. The similarity determining section compares the sound characteristic parameter obtained from said sound characteristic parameter analyzing section and the retrieving key sound characteristic parameter obtained from said retrieving key sound parameter reading section and determines the similarity of them to retrieve a part of the audio signal to be analyzed that resembles the audio signal operating as retrieving key and obtained from the sound source of the retrieving key.

Abstract: A detection circuit of an electronic device, includes a jack circuit, a first switch, a second switch, and a detection end. The jack circuit includes a type detection pin and a connection detection pin. The first switch includes a first control end. The connection detection pin is connected to the first control end to control the first switch on or off according to whether the audio device is connected to the electronic device. The second switch is connected to ground via the first switch. The second switch includes a second control end. The type detection pin is connected to the second control end to control the second switch on or off according to the type of the audio device. The detection end is connected to the first switch via a first resistor, and is connected to the second switch via a second resistor. The detection end is adapted to generate different signals to indicate a connection status of the electronic device and the audio device.

Abstract: An sound reproducing apparatus is configured such that a correction amount is calculated on the basis of a masking model, determiner for determining a correction parameter correcting input signal so as to be natural for human ears to hear on the basis of the calculated correction amount is provided, and a characteristic of correction filter is changed on the basis of a result of determiner when inputting vehicle speed information.

Abstract: Disclosed are a sound quality display apparatus, a sound quality display method, a computer readable medium on which a sound quality display program is recorded, and a sound camera. The apparatus includes a sound detector detecting sound generated from a sound source, a background photographing unit photographing background where the sound source is positioned, a sound source signal generator creating a sound source signal as a sound signal at a sound source plane where the sound source is positioned by analyzing the sound signal detected by the sound detector, a sound quality data generator creating sound quality data at the sound source plane by processing of the sound source signal, and a display displaying sound quality image data created by overlaying image data of the background photographed by the background photographing unit with the sound quality data.

Abstract: A device for monitoring outside sound through a closed window includes a housing positioned inside the window. The housing contains a filter, a squelch circuit, an amplifier, and a speaker. A microphone is positioned outside the window and connected to the housing. The housing includes a volume control, a sensitivity control, and a mode control for bypassing the squelch circuit.

Abstract: A tone detection device for detecting whether an input signal having a tone. The device includes a volume gain calculation unit performing a volume gain treatment on frame data and outputting the volume-gain-treated frame data and energy in time domain thereof, a threshold calculation unit calculating a threshold value on the basis of the energy of the volume-gain-treated frame data, a filter transforming the volume-gain-treated frame data by an algorithm and outputting a characteristic value in a first period, and a comparator comparing the first characteristic value and the threshold value to generate a result and determining the frame data of the input signal has the tone on the basis of the result.

Abstract: A system for monitoring the operation of one or more devices is provided. The system includes a microphone acoustically coupled to one the monitored device. An analog-to-digital-converter samples the microphone and a processor examines the resultant digital signal for the occurrence of an abnormal event.

Abstract: A result of a sound source direction measurement based on an output of an REMA (first microphone array) (11) and a result of a sound source position measurement based on an output of an IRMA (second microphone array) (12) are integrated through a particle filter or in space. Thus, the different microphones, i.e., the REMA (11) and the IRMA (12) can cancel mutual defects or ambiguities with each other. Therefore, from views of improvement in accuracy and robustness a performance of sound source localization can be improved.