Abstract: A multi-channel audio system having multiple loudspeakers is used to obtain information on the location of one or more independent noise sources within an area covered by the loudspeakers. Within the multi-channel audio system, an audio output device has an input for coupling to and receiving audio signals from one or more audio sources; an audio processing module for generating a audio drive signals and providing them on respective outputs to a number of loudspeakers. A sensing module has inputs connected to respective outputs of the audio processing module, for receiving signals corresponding to sound sensed by the loudspeakers. The sensing module includes a discriminator for discriminating between signals corresponding to the audio drive signals and sensed signals from an independent noise source within range of the loudspeakers. A position computation module determines a two or three dimensional position of each independent noise source sensed, relative to the loudspeakers.

Abstract: A hands-free loudspeaker system which is capable of achieving high-quality voice amplification without requiring a human speaker to move to a microphone or a microphone to be moved to a human speaker. A microphone whose input level has continued to be above a threshold value for not shorter than a predetermined time period is detected, based on input signals from dispersedly arranged microphones. An input signal from the microphone is selected and outputted to a loudspeaker at an output level or with a delay time, according to a location of the loudspeaker. A preset lowest threshold level is initially set to the threshold value, and an input level of the microphone higher than the threshold value is newly set to the same, while when the input level is lower than the threshold value, a lower value is set to the same in a step-by-step manner.

Abstract: An exemplary media player system includes a media player and an earphone. The earphone includes a first earpiece, a second earpiece, a primary volume control unit, and a secondary volume control unit. The first earpiece and the second earpiece are configured for receiving first channel audio signals and second channel audio signals respectively, and converting the audio signals to yield audible sounds. The primary volume control unit adjusts the volume levels of the first earpiece and the second earpiece simultaneously. The secondary volume control unit adjusts the volume of sounds outputted from one of the first earpiece and the second earpiece.

Abstract: An interactive assembly for displaying a sequence of words from a text comprising: a processor; at least one memory operatively connected to the processor; at least one speech recognizer operatively connected to the processor; at least one audio input operatively connected the speech recognizer; at least one display operatively connected to the speech recognizer for displaying words of a song based upon the words recognized by the speech recognizer; whereby based upon a word received by the audio input, the next words to be recited incrementally displayed onto the at least one display contemporaneously with the input received from the operator. A method for interactively display the words of a text sequentially on a display comprising: receiving an inputted vocal input in an audio input device; inputting the inputted vocal input into a speech recognizer; comparing the inputted vocal input with words stored in sequence in memory; outputting on a display screen only the next sequential words for display.

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: In an information detecting apparatus (1), a speech kind discrimination unit (11) discriminates and classifies an audio signal at an information source into kind (category) such as music or speech, etc. on a predetermined time basis, and a memory unit/recording medium (13) records discrimination information thereof. A discrimination frequency calculating unit (15) calculates, on a predetermined time basis, discrimination frequency every kind at a predetermined time period longer than the time unit.

Abstract: A multiplayer participation type gaming system, comprising: a plurality of gaming machines arranged on a predetermined play area; and a sound insulating wall disposed between adjacent gaming machines among the gaming machines, each of the gaming machines carries out the processing of: performing an arithmetic operation, in response to a play of the player, on a value indicative of the game result, and storing after updating the play history data stored in the memory by using the result of the arithmetic operation; comparing the updated play history data with the predetermined threshold value data; generating, when it is determined that the play history data exceeds the threshold value data, sound data based on the play history data by using the plurality of voice generating original data stored in the memory; and outputting a sound from the speaker based on the sound data thus generated.

Abstract: Embodiments of the invention address the deficiencies of the prior art by providing a method, apparatus, and program product to cooperatively mediate between voice-enabled operations and business logic. The method comprises receiving XML data and generating at least one object from the XML data. The method further comprises, in response to determining that the at least one object has been called, implementing an operation defined by a portion of the object.

Abstract: An electronic appliance includes a microphone which detects a clapping sound, an edge signal extractor, an edge pulse generator and a judgment processing circuit. The judgment processing circuit generates a first gate having a first time width to detect whether or not a second clapping sound has been generated after elapse of a first predetermined time from a first time when the microphone detects a first clapping sound and the edge pulse generator generates the first edge pulse corresponding to the first clapping sound. Subsequently, the judgment processing circuit generates a second gate having a second time width to detect whether or not a third clapping sound has been generated after elapse of a second predetermined time from a second time when the edge pulse generator generates the second edge pulse corresponding to the second clapping sound.

Abstract: A method and an apparatus for detecting audio signals are disclosed. The input audio signal is inspected to check whether it is a foreground frame or a background frame; the detected background signal is further inspected according to the music eigenvalue and the decision rule. Therefore, background music can be detected, and the classifying performance of the voice/music classifier is improved.

Abstract: An audio network system that performs transport of audio signals among nodes by cascading a plurality of nodes each including two sets of transmission I/Fs and reception I/Fs, and circulating among the nodes in each fixed period an audio transport frame generated by a master node, is configured such that the master node generates the audio transport frame in an S-th period based on the audio transport frame in an (S?k)-th period, and each of the other nodes delays the audio signals written by another node after the audio transport frame is generated by the master node until the audio transport frame is transmitted to the self node, by k period(s) with respect to the other audio signals, for use in signal processing.

Abstract: A method of extracting a voice command produced in an enclosed or partially enclosed environment, includes providing an impulse response signal of the enclosed or partially enclosed environment; recording the voice command and ambient sounds; and using the impulse response signal to extract the recorded voice command.

Abstract: Digital process for authentication of a user of a database for access to protected data or a service reserved for a defined circle of users or for the use of data currently entered by the user, wherein a voice sample currently enunciated during an access attempt by the user is routed to a voice analysis unit and, herein, a current voice profile is computed and this is compared in a voice profile comparison unit against a previously stored initial voice profile and, in response to a positive comparison result, the user is authenticated and a first control signal enabling access, but in response to a negative comparison result a second control signal disabling access or triggering a substitute authentication procedure is generated.

Abstract: A VOX interface is provided that interfaces with at least two communication paths to two separate devices. One communication path may be to a first electronic device, such as a central monitoring station remote from the VOX interface, and another communication path may be to one or more remote devices, such as to one or more patient devices remote from the VOX interface. The VOX interface may determine which of the communication paths is sending a signal (such as an audio signal), and configure the VOX interface (via one or more switches) to pass the signal through. Specifically, the VOX interface may sense signals indicating the presence of audio from a central station audio bus and the patient station bus, may do some background noise filtering on the signals, and may pass these filtered values through a differentiator circuit to determine which bus presented the audio.

Abstract: A small array microphone system includes an array microphone having a plurality of microphones and operative to provide a plurality of received signals, each microphone providing one received signal. A first voice activity detector (VAD) provides a first voice detection signal generated using the plurality of received signals to indicate the presence or absence of in-beam desired speech. A second VAD provides a second voice detection signal generated using the plurality of received signals to indicate the presence or absence of out-of-beam noise when in-beam desired speech is absent. A reference signal generator provides a reference signal based on the first voice detection signal, the plurality of received signals, and a beamformed signal, wherein the reference signal has the desired speech suppressed. A beamformer provides the beamformed signal based on the second voice detection signal, the reference signal, and the plurality of received signals, wherein the beamformed signal has noise suppressed.

Abstract: Provided is a method and system for audio privacy that includes receiving a first sound signal at a microphone proximal to a user's ear, generating a second sound signal based on the first sound signal and a stored filter, the second sound signal interfering with the first sound signal, and emitting the second sound signal from a speaker proximal to the user's ear.

Abstract: An impulsive communication activated device for connection to a computer input port for controlling a computer, is disclosed. The disclosed device includes an output connector for communicating with the computer input port, and includes a sensor for detecting an impulsive communication such as a hand clap or other predetermined input, to generate an impulsive signal. The disclosed device includes a comparator for responding to both a reference signal and to the impulsive signal, and to determine if the impulsive signal differs substantially from the reference signal to generate a start signal. The disclosed device includes a logic circuit responsive to the start signal for generating a predetermined computer command signal to control the computer.

Abstract: A signal classifying method and apparatus are disclosed. The signal classifying method includes: obtaining a spectrum fluctuation parameter of a current signal frame determined as a foreground frame, and buffering the spectrum fluctuation parameter; obtaining a spectrum fluctuation variance of the current signal frame according to spectrum fluctuation parameters of all buffered signal frames, and buffering the spectrum fluctuation variance; and calculating a ratio of signal frames whose spectrum fluctuation variance is above or equal to a first threshold to all the buffered signal frames, and determining the current signal frame as a speech frame if the ratio is above or equal to a second threshold or determining the current signal frame as a music frame if the ratio is below the second threshold.

Abstract: A method and an apparatus for detecting audio signals are disclosed. The input audio signal is detected to determine whether it is a background frame. The detected background signal is further detected according to a music characterization value and a decision rule. Therefore, background music can be detected, and the classifying performance of the voice/music classifier is improved.

Abstract: There is provided a voice input support program, voice input support device, and voice input support method for assisting users to input voice properly. A voice input support program executable from a computer, for assisting voice input under a noisy environment by executing: a noise measurement step that obtains a measured noise level value by measuring a noise level; and a notification step that outputs an instruction about at least one of a voice level at which voice should be spoken and speak timing, based on at least one registered voice level reference value and the measured noise level value measured by the noise measurement step.

Abstract: A system, set-top box, and method of providing audio content are disclosed. In a particular embodiment, the set-top box device includes an audio input device that is configured to receive a voice command. The set-top box device also includes a display disposed on a visible side of a set-top box housing. Additionally, the set-top box device includes a media control logic module to retrieve audio content identified by the voice command and data related to the audio content. The media control logic module transmits an audio signal corresponding to the audio content to an audio output device. The data related to the audio content is displayable on the display.

Abstract: A voice acquisition system for a vehicle includes an interior rearview mirror assembly. The mirror assembly may include a microphone for receiving audio signals within a cabin of the vehicle and generating an output indicative of these audio signals. The microphone may provide sound capture for a hands free cell phone system, an audio recording system and/or an emergency communication system. The system may include a control that is responsive to the output from the microphone and that distinguishes vocal signals from non-vocal signals present in the output. The microphone may provide sound capture for at least one accessory of the equipped vehicle, and the accessory may be responsive to a vocal signal captured by the microphone. The interior rearview mirror assembly may include at least one accessory, such as an antenna, a video device, a security system status indicator, a tire pressure indicator display and/or a loudspeaker.

Abstract: A voice recognition system and method thereof. The communication system comprises a setup controller, a voice recognition controller, and an application controller. The setup controller receives a voice keyword table including a voice recognizable keyword and a corresponding application instruction. The voice recognition controller, coupled to the setup controller, receives the voice keyword table from the setup controller, receives a first voice input, and matches a first voice input to the voice recognizable keyword to determine the corresponding application instruction. The application controller, coupled to the voice recognition controller, receives and performs the corresponding application instruction. The setup controller, the voice recognition controller, and the application controller communicate through wireless communication.

Abstract: Ear-level full duplex audio communication systems each include one or two ear attachment devices, such as in-the-ear (ITE) or behind-the-ear (BTE) devices, that wirelessly communicates to a remote device such as a computer, a personal digital assistant (PDA), a cellular phone, a walkie talkie, or a language translator. When used as a hearing aid, such a system allows a hearing impaired individual to communicate with or through the remote device, such as to talk to another person through a cellular phone. When being used as an ear piece wirelessly extended from the remote device, such system allows an individual with normal hearing to privately communicate with or through the remote device without the need of holding the device or wearing any device wired to the remote device. Each ear attachment device includes a voice operated exchange (VOX), housed within the device, to preserve energy and hence, maximize the period between battery replacement or recharges.

Abstract: According to one embodiment, various feature parameters are calculated for distinguishing between a speech and music and between music and background sound for an input audio signal. With the feature parameters, score determination is made as to whether the input audio signal is close to a speech signal or a music signal. If the input audio signal is determined to be close to music, the preceding score determination result is corrected considering the influence of background sound. Based on the corrected score value, a sound quality correction process for a speech or music is applied to the input audio signal.

Abstract: In one embodiment, the present invention is a method and apparatus for active noise cancellation. In one embodiment, a method for recognizing user speech in an audio signal received by a media system (where the audio signal includes user speech and ambient audio output produced by the media system and/or other devices) includes canceling portions of the audio signal associated with the ambient audio output and applying speech recognition processing to an uncancelled remainder of the audio signal.

Abstract: An electronic processing apparatus comprises one or more remote interactive display devices and a calculator unit, wherein the display device comprises at least a screen, a connector selectively connectable to the calculator unit, a user interface device and a microprocessor, whereas the calculator unit comprises at least one or more connectors, able to be selectively connected to the connector, a processor and a non-volatile memory. There is also a single connection element able to connect the connector and a selected one of the connectors of the calculator unit, both for data exchange and also for the transfer of energy for electric feed.

Abstract: The present invention provides elimination of atmosphere that is not suitable to a space concerned by controlling atmosphere. The present invention is characterized that atmosphere in space is analyzed based on voice, and if atmosphere that is not suitable to the space is detected, the atmosphere in the space is controlled by choosing and irradiating illumination that enables to create atmosphere suitable to the space, atmosphere in the space is controlled for creating atmosphere that is suitable to the space.

Abstract: In order to monitor various types of noises which are to be introduced on signals through signal lines on a circuit board and automatically adjust the thresholds for signal state discriminations to make it possible to surely make a signal state discrimination without being affected by these noises even if the amplitude of a signal is reduced for higher-speed transmission and lowered electric power, there is provided a configuration comprising a signal generation unit generating a noise monitor signal; a noise monitor signal line receiving and propagating the noise monitor signal; a noise detection unit detecting a noise which has been introduced into that noise monitor signal propagated through the noise monitor signal line and which affects a state discrimination using a threshold; and a threshold adjustment unit, if the noise detection unit detects the noise, adjusting the threshold such that the state discrimination is not affected by the noise.

Abstract: A voice-activated, or voice-operated, remote control is adapted to reduce or eliminate the sound output of one or more entertainment appliances in order quiet the environment and thereby improve the effectiveness and accuracy its voice recognition functions. In one aspect of the present invention, the remote control sends a mute command to an appliance under control and listens for a voice command during an ensuing predetermined time period. In a further aspect of the present invention, the remote control determines which one of a plurality of unmute sequences is to be transmitted.

Abstract: In accordance with one aspect of the dynamic volume control, an indication that a user desires to input oral data to a system through one or more microphones of the system is received. In response to receipt of the indication, a volume level for audible signals output by one or more speakers of the system is automatically adjusted. In accordance with another aspect of the dynamic volume control, an indication that a communications source is about to output data through one or more speakers of a system is received. In response to receipt of the indication, a volume level for audible signals output by the one or more speakers is automatically adjusted based at least in part on a current volume setting. The volume level for the audible signals can be determined based on one or more of a variety of different parameters.

Abstract: A hands-free loudspeaker system which is capable of achieving high-quality voice amplification without requiring a human speaker to move to a microphone or a microphone to be moved to a human speaker. A microphone whose input level has continued to be above a threshold value for not shorter than a predetermined time period is detected, based on input signals from dispersedly arranged microphones. An input signal from the microphone is selected and outputted to each loudspeaker at an output level or with a delay time, according to a location of the loudspeaker. A preset lowest threshold level is initially set to the threshold value, and an input level of the microphone higher than the threshold value is newly set to the same, while when the input level is lower than the threshold value, a lower value is set to the same in a step-by-step manner.

Abstract: According to one embodiment, sound quality control processing for speech or music is performed by calculating various kinds of characteristic parameters to determine a speech signal and a music signal from an input audio signal and determining the input audio signal closer to the speech signal or music signal based on a score difference between a sum of scores provided to characteristic parameters indicating the speech signal and that of scores provided to characteristic parameters indicating the music signal.

Abstract: In a mobile phone terminal, in which a sound, such as the audio, may be synchronized with a moving picture even in case a sound processor unit processing the sound, such as audio, is controlled by a controller differing from the controller controlling a moving picture processing unit processing the moving picture, there is provided a synchronization controller which calculates the transmission time of a signal transmitted between an audio processing CPU and a moving picture processing CPU, transmits an audio processing start signal, commanding the start of the audio processing, to the audio processing CPU through the moving picture processing CPU, and transmits, at a timing when the previously calculated transmission time has elapsed as from the transmission of the sound processing start signal, a moving picture processing start command signal, commanding the start of the moving picture processing, to the moving picture processing CPU.

Abstract: An interface which allows a surgeon to operate multiple surgical devices from a single input device. The input device may be a foot pedal that provides output signals to actuate a number of different surgical devices. The surgical devices may include a robotic arm, a laser, an electrocautery device, or an operating table. The interface has an input channel that is coupled to the input device and a plurality of output channels that are coupled to the surgical devices. The interface also has a select input channel which can receive input commands to switch the input channel to one of the output channels. The select channel may be coupled to a speech interface that allows the surgeon to select one of the surgical devices with a voice command. The surgeon can operate any device by providing an input command which switches the input channel to the desired output channel.

Abstract: In one embodiment, one or more users may be participating in a conversation. In one example, a first user may be speaking into a speaker end device and a second user may be listening at a listener end device. The second user may be in an environment where noise may be present. Particular embodiments determine characteristics of the noise at the listener end device. Characteristics of a voice signature for a user speaking with the speaker end device are also determined. Comprehension enhancement of voice signals received from speaker end device is then performed based on characteristics of the noise at the listener end device and characteristics of the voice signature. For example, the signature of the voice signals may be altered to lessen the overlap with the noise.

Abstract: Various embodiments reduce noise within a particular environment, while isolating and capturing speech in a manner that allows operation within an otherwise noisy environment. In one embodiment, an array of one or more microphones is used to selectively eliminate noise emanating from known, generally fixed locations, and pass signals from a pre-specified region or regions with reduced distortion.

Abstract: In the method according to the invention a signal processing unit receives signals from at least two microphones worn on the user's head, which are processed so as to distinguish as well as possible between the sound from the user's mouth and sounds originating from other sources. The distinction is based on the specific characteristics of the sound field produced by own voice, e.g. near-field effects (proximity, reactive intensity) or the symmetry of the mouth with respect to the user's head.

Abstract: The present invention provides elimination of atmosphere that is not suitable to a space concerned by controlling atmosphere. The present invention is characterized that atmosphere in space is analyzed based on voice, and if atmosphere that is not suitable to the space is detected, the atmosphere in the space is controlled by choosing and irradiating illumination that enables to create atmosphere suitable to the space, atmosphere in the space is controlled for creating atmosphere that is suitable to the space.

Abstract: A system is provided for controlling personalized settings in a vehicle. The system includes a microphone for receiving spoken commands from a person in the vehicle, a location recognizer for identifying location of the speaker, and an identity recognizer for identifying the identity of the speaker. The system also includes a speech recognizer for recognizing the received spoken commands. The system further includes a controller for processing the identified location, identity and commands of the speaker. The controller controls one or more feature settings based on the identified location, identified identity and recognized spoken commands of the speaker. The system also optimizes on the beamforming microphone array used in the vehicle.

Abstract: A remote control device for wirelessly controlling a second device is disclosed that allows the user to remotely control the second device using spoken commands. The remote control device is preferably dimensioned to be useful as a handheld device. The remote control device includes one or more microphones into which the user can speak a command. The remote control device also includes a transmitter, for example a radio frequency transmitter and/or an optical transmitter, for transmitting a signal to the second device based on the user's spoken command.

Abstract: A steambath system is responsive to verbal commands to control the various aspects of the shower and steam environment such as water flow rate, water temperature, steam activation/de-activation, lighting intensity and/or color, music volume, music content, quantity and type of fragrance released, showerhead selection, radio activation, volume and station selection, television activation, volume and channel selection, DVD player activation and DVD selection, CD player activation and CD selection, and humidity control.

Abstract: At least one exemplary embodiment is directed to a method and/or a device for voice operated control. The method can include method measuring an ambient sound received from at least one Ambient Sound Microphone, measuring an internal sound received from at least one Ear Canal Microphone, detecting a spoken voice from a wearer of the earpiece based on an analysis of the ambient sound and the internal sound, and controlling at least one voice operation of the earpiece if the presence of spoken voice is detected. The analysis can be a non-difference comparison such as a correlation analysis, a cross-correlation analysis, and a coherence analysis.

Abstract: An earpiece (100) and a method (300) personalized voice operable control can include capturing (302) an ambient sound from an Ambient Sound Microphone (111) to produce an electronic ambient signal (426), delivering (304) audio content (402) to an ear canal (131) by way of an Ear Canal Receiver (125) to produce an acoustic audio content (404), capturing (306) in the ear canal an internal sound (402) from an Ear Canal Microphone (123) to produce an electronic internal signal (410), wherein the electronic internal signal includes an echo of the acoustic audio content and a spoken voice generated by a wearer of the earpiece, detecting (312) the spoken voice in the electronic internal signal in the presence of the echo, and controlling (314) a voice operation of the earpiece when the spoken voice is detected.

Abstract: An edge signal extractor generates an edge signal according to a sound wave generated for controlling the electronic appliance. According to the edge signal, an edge pulse generator generates an edge pulse. If a predetermined number of such edge signals are generated, a controller displays, on a display of the electronic appliance, one of menu images associated with control operations of the electronic appliance in an active state. The controller alternately displays, in an active state, the menu images on the display at predetermined intervals. When the edge pulse is generated, the controller carries out the control operation associated with the presently displayed menu image with respect to an electronic appliance.

Abstract: A voice acquisition system for a vehicle includes an interior rearview mirror assembly attached at an inner portion of the vehicle. The mirror assembly preferably includes at least one microphone for receiving audio signals within a cabin of the vehicle and generating an output signal indicative of these audio signals, the at least one microphone providing sound capture for at least one of a hands free cell phone system, an audio recording system and an emergency communication system. The system includes a control that receives the output signal from the at least one microphone and distinguishes the presence of vocal signals from non-vocal signals. The system preferably comprises a learning mode whereby a vocal characteristic of a particular driver or occupant of the vehicle is learnt so that the ratio of vocal signals emanating from that particular driver or occupant to non-vocal noise signals received by the system is enhanced.

Abstract: A method (100) of dynamically assigning a plurality of voice ports and arranging a plurality of menu choice prompting sequences for at least one application includes the step of receiving (102) a plurality of incoming calls and monitoring (104) at least one among a recognition error rate for each menu option and execution path, a success rate for completing a desired transaction and an associated origination number, and monitoring frequently used menu options. In response to the monitoring steps, at least one of the menu choice prompting sequences among the plurality of menu choice prompting sequences can be rearranged (110) and at least one voice port among the plurality of voice ports can be reassigned (114).

Abstract: At least one exemplary embodiment is directed to a method and/or a device for voice operated control. The method can include method measuring an ambient sound received from at least one Ambient Sound Microphone, measuring an internal sound received from at least one Ear Canal Microphone, detecting a spoken voice from a wearer of the earpiece based on an analysis of the ambient sound and the internal sound, and controlling at least one voice operation of the earpiece if the presence of spoken voice is detected. The analysis can be a sound pressure level (SPL) difference, a correlation, a coherence, and a spectral difference.

Abstract: An ultrasound imaging system includes a direction-tracking microphone that is able to determine the direction of a voice command and to cause the microphone to selectively receive acoustic inputs from the determined direction. A voice recognition then interprets the voice command and controls the operation of the ultrasound imaging system accordingly. The direction tracking microphone may, for example, select one of several unidirectional microphones that receives the loudest signal or a phased array of omnidirectional microphones.

Abstract: An apparatus for virtual navigation and voice processing is provided. A system that incorporates teachings of the present disclosure may include, for example, a computer readable storage medium having computer instructions for processing voice signals captured from a microphone array, detecting a location of an object in a touchless sensory field of the microphone array, and receiving information from a user interface in accordance with the location and voice signals.