Abstract: An audio device may be configured to produce output audio and to also capture input audio for speech recognition. Beamforming may be used to produce directional audio signals that emphasize sounds received from different directions. Acoustic echo cancellation (AEC) may be used to remove components of output audio from the directional input signals. AEC may be implemented by an adaptive filter based on dynamically optimized filter coefficients. The filter coefficients may be analyzed to detect the presence of potentially problematic reflective surfaces near the audio device. When such a reflective surface is detected, the user may be notified by illuminating a directional indicator on a housing of the audio device, where the directional indicator indicates the direction of the potentially problematic surface relative to the audio device.

Abstract: Provided is a remote control comprising a housing, a keyboard (20) mounted on the housing, a processor (106) electrically connected to the keyboard (20), an analog-digital converter (103) and a high-frequency wireless transmission unit (108) electrically connected to the processor (106), a sampling circuit (102) electrically connected to the analog-digital converter (103) and a microphone interface (30) electrically connected to the sampling circuit (102). Also disclosed is a television system. The remote control and television system in the present solution are simple to operate and can satisfy user's multi-function requirements.

Abstract: A security system can be used to trigger appliances. The security system can comprise a light socket camera that can be rotatably attached to a light socket of a building. The light socket camera can receive an audible instruction from a user, and in response to receiving the audible instruction from the user, the security system can trigger the appliance to perform an operation.

Abstract: A signal processing apparatus feeding a frame of a signal in frequency domain of a reception voice signal into a sound echo canceler includes a first reception section for receiving frames of the reception voice signal in frequency domain before having a rate-of-speech change process applied; a second reception section for receiving frames of a signal in time domain having the rate-of-speech change process applied by units of frames; and a frequency-domain frame synthesis section for synthesizing a frame of the signal in frequency domain of the reception voice signal based on the signal in time domain having the rate-of-speech change process applied at a frame currently being processed by the signal processing apparatus, and a frame of the reception voice signal in frequency domain corresponding to the signal in time domain having the rate-of-speech change process applied.

Abstract: The present invention discloses a flat speaker output device and a method for starting the same. Wherein, a plurality of flat speakers utilizes an initial delay unit and a plurality of intermediary delay units connected in series. The initial delay unit connects with the power controller and a first one of the flat speakers. The intermediary delay units respectively connect with the residual each flat speakers. The power controller controls a power source to the initial delay unit to delay the start of the first one of flat speakers, and outputs the power source to the intermediary delay units to sequentially delay the starts time of the residual each flat speaker. The present invention can sequentially start flat speakers without using a high-output power supply device and thus decrease the required capacity of the external power supply device.

Abstract: Remote surveillance systems can include an outward portion that faces away from a power outlet and can include an inward portion with electrical prongs that protrude into the power outlet. The power outlet can mechanically support the inward portion and a camera assembly. The remote surveillance system can take videos using the camera and can send the videos to a remotely located computing device.

Abstract: A system and method of operating an automatic speech recognition application over an Internet Protocol network is disclosed. The ASR application communicates over a packet network such as an Internet Protocol network or a wireless network. A grammar for recognizing received speech from a user over the IP network is selected from a plurality of grammars according to a user-selected application. A server receives information representing speech over the IP network, performs speech recognition using the selected grammar, and returns information based upon the recognized speech. Sub-grammars may be included within the grammar to recognize speech from sub-portions of a dialog with the user.

Abstract: Individual adjustment of audio volume and video properties in a computer network conference environment is provided. For audio adjustment, a buffer collects incoming streams; a stream decoder decodes the buffered audio streams, a gain adjustment applies a gain increase or decrease to the individual audio stream, and a mixer combines each of the individual signals together. The gain adjustment module receives input from a user interface control associated with each participant, and adjusts the volume of that participant's stream accordingly. If a requested increase in gain would cause an overflow of the signal, only a gain increase that will avoid such overflow is applied. Video properties such as brightness, contrast and saturation are also adjustable. Properties of the user's transmitted audio and video streams are also adjustable.

Abstract: According to one embodiment, an audio controlling apparatus includes a first receiver configured to receive audio signal, a second receiver configured to receive environmental sound, a temporary gain calculator configured to calculate temporary gain based on environmental sound received by second receiver, a sound type determination module configured to determine sound type of main component of audio signal received by first receiver, and a gain controller configured to stabilize temporary gain that is calculated by temporary gain calculator and set gain, when it is determined that sound type of main component of audio signal received by first receiver is music.

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: A method for interpreting messages, user-defined alert conditions, voice commands and performing an action in response is described. A method for annotating media content is described. A method for presenting additional content associated with media content identified based on a fingerprint is described. A method for identifying that an advertisement portion of media content is being played based on a fingerprint derived from the media content is described. A method of one media device recording particular media content automatically in response to another media device recording the particular media content is described. A method of concurrently playing media content on multiple devices is described. A method of publishing information associated with recording of media content is described. A method of deriving fingerprints by media devices that meet an idleness criteria is described. A method of loading, modifying, and displaying a high definition frame from a frame buffer is described.

Abstract: A portable system for enhancing communication between users in proximity to each other while listening to a common audio source includes headsets with an electroacoustic transducer for providing sound to a respective user's ear, and a voice microphone for detecting sound of the respective user's voice and providing a microphone input signal, and an electronic device integral to the first headset and in communication with the second headset. The electronic device generates a side-tone signal based on the microphone input signal from the first headset, generates a voice output signal based on the microphone input signal from the first headset, receives a content input signal, combines the side-tone signal with the content input signal and a far-end voice signal associated with the second headset to generate a combined output signal, and provides the combined output signal to the first headset for output by the first headset's transducer.

Abstract: An electronic device includes a microphone, a bias-supply device and a voice-recognition device. The bias-supply device is configured to provide a first bias voltage to serve as an operation voltage of the microphone, when the electronic device is operated in a power-saving mode, such that the microphone transforms a voice signal into a first output signal. The voice-recognition device is configured to receive the first output signal and output a control signal, when the first output signal has a predetermined signal, to enable the electronic device be operated in a normal operation mode and the bias-supply device to provide a second bias voltage that is higher than the first bias voltage to serve as the operation voltage of the microphone, such that the microphone transforms the voice signal into a second output signal and outputs the second signal to a core circuit.

Abstract: A method of controlling a function of an acoustic detector includes storing a reference power spectrum of a reference acoustic signal, the reference power spectrum being associated with a mode of operation of the acoustic detector; receiving an acoustic sound signal, the acoustic sound signal including a tone and harmonic tones; generating a real-time acoustic power spectrum of the acoustic sound signal responsive to the receiving of the acoustic sound signal; and performing the mode of operation when a comparison of the real-time acoustic power spectrum with the reference acoustic power spectrum indicates that the acoustic sound signal is a control signal.

Abstract: Improvements in voice signals transmitted within communication systems are obtained by use of adaptive filters, front and rear microphones, noise cancelling systems and other means and methods. Disclosed embodiments include the use of directional microphones, primary inputs, secondary inputs, adaptive weight generators, canceller outputs to improve signal to noise ratios and other communication attributes.

Abstract: A method and apparatus are provided for processing a set of communicated signals associated with a set of muscles, such as the muscles near the larynx of the person, or any other muscles the person use to achieve a desired response. The method includes the steps of attaching a single integrated sensor, for example, near the throat of the person proximate to the larynx and detecting an electrical signal through the sensor. The method further includes the steps of extracting features from the detected electrical signal and continuously transforming them into speech sounds without the need for further modulation. The method also includes comparing the extracted features to a set of prototype features and selecting a prototype feature of the set of prototype features providing a smallest relative difference.

Abstract: A system for simultaneous wireless control of lights and motorized window coverings that include sheer interior shade material and blackout exterior shade material. The system includes a voice activation module which is wirelessly connected to a plurality of motorized window coverings using a first transceiver and a first wireless protocol and a plurality of lights using a second transceiver and a second wireless protocol. The voice activation module is preprogrammed to listen for a trigger phrase followed by any one of a number commands. Once the trigger phrase and a command are recognized, the wireless device transmits an over-the-air signal to the plurality of motorized window coverings and lights. The wireless device also includes an indicator that provides a visual or audible indication to the user that the trigger phrase, and/or a command has been recognized.

Abstract: An earpiece (100) is provided. The earpiece can include an Ambient Sound Microphone (111) configured to capture ambient sound, an Ear Canal Microphone (123) configured to capture internal sound in the ear canal, a memory (208) configured to record at least a portion of the history of the ambient sound and the internal sound, and a processor (121) configured to save a recent portion of the history responsive to an event.

Abstract: An earpiece (100) and a method (640) for acoustic management of multiple microphones is provided. The method can include capturing an ambient acoustic signal from an Ambient Sound Microphone (ASM) to produce an electronic ambient signal, capturing in an ear canal an internal sound from an Ear Canal Microphone (ECM) to produce an electronic internal signal, measuring a background noise signal, and mixing the electronic ambient signal with the electronic internal signal in a ratio dependent on the background noise signal to produce a mixed signal. The mixing can adjust an internal gain of the electronic internal signal and an external gain of the electronic ambient signal based on the background noise characteristics. The mixing can account for an acoustic attenuation level and an audio content level of the earpiece.

Abstract: A method for controlling voice activation by a target keyword in a mobile device is disclosed. The method includes receiving an input sound stream. When the input sound stream indicates speech, the voice activation unit is activated to detect the target keyword and at least one sound feature is extracted from the input sound stream. Further, the method includes deactivating the voice activation unit when the at least one sound feature indicates a non-target keyword.

Abstract: A method and apparatus to evaluate a quality of an audio signal, in which the number of effective channels is determined for each of a reference signal of a current frame and a test signal indicative of the reference signal that has passed through an audio codec, and an audio quality evaluation score of the current frame is calculated by evaluating an audio quality of the current frame based on the determined number of effective channels for each of the reference signal and the test signal by means of a predetermined evaluator.

Abstract: A headset having game, chat and microphone audio signals is provided with a programmable signal processor for individually modifying the audio signals and a memory configured to store a plurality of user-selectable signal-processing parameter settings that determine the manner in which the audio signals will be altered by the signal processor. The parameter settings collectively form a preset, and one or more user-operable controls can select and activate a preset from the plurality of presets stored in memory. The parameters stored in the selected preset can be loaded into the signal processor such that the sound characteristics of the audio paths are modified in accordance with the parameter settings in the selected preset.

Abstract: A terminal and a method are disclosed. The method utilizes a processor for executing an object using a voice command, and the method includes: searching for an executable object in an application; and associating a text command information with the executable object. The method can further include: receiving a voice; recognizing the voice to obtain a voice command; and executing the executable object, when the voice command is similar to the text command information associated with the executable object.

Abstract: Devices and techniques for speech detection using low power microelectrical mechanical systems (MEMS) sensor are described, including monitoring acoustic energy using a microelectrical mechanical system sensor, detecting a presence of speech using a voice activity detection device comprising a voice activity detection logic and the microelectrical mechanical system sensor formed on die, switching a host system from a first power mode to a second power mode, using a power manager, upon receiving a signal from the voice activity detection device indicating a presence of speech, the host system comprising one or more sensors and a speech recognition module configured to recognize a speech command, and taking an action in response to the speech command.

Abstract: Devices and techniques for speech detection using low power microelectrical mechanical systems (MEMS) sensor are described, including a power source, a voice activity detection device connected to the power source and having a microelectrical mechanical system sensor formed on die with a digital signal processor and a voice activity detection logic, and a host system connected to the power source and the voice activity detection device, the host system having sensors, a power manager configured to control power being consumed by the host system according to various power modes, and a speech recognition module, where the voice activity detection device is configured to provide a signal to the host system indicating the presence of speech.

Abstract: A method includes receiving input from a remote control device at a set-top box device. The input identifies media content that includes audio content and video content. The input also indicates that the audio content is to be output via an audio output device coupled to the set-top box device without sending the video output from the set-top box device to an external display device coupled to the set-top box device. The method includes determining whether the media content is stored at a local memory device of the set-top box device. When the media content is stored at the local memory device, the method includes retrieving the media content from the local memory, sending a first audio signal corresponding to the audio content of the media content to the audio output device and blocking output of the video content to the external display device.

Abstract: A method for navigating among tracks uses limited input commands. A plurality of tracks is associated with a currently playing queue. One of the tracks in the queue is identified for a current focus. A voice command is received that includes a directional aspect and a metadata filtering aspect. In response, the focus moves to a second track having a different value for the selected metadata filtering category.

Abstract: A speech signal processing device is equipped with a power acquisition unit, a probability distribution acquisition unit, and a correspondence degree determination unit. The power acquisition unit accepts an inputted speech signal and, based on the accepted speech signal, acquires power representing the intensity of a speech sound represented by the speech signal. The probability distribution acquisition unit acquires a probability distribution using the intensity of the power acquired by the power acquisition unit as a random variable. The correspondence degree determination unit determines whether a correspondence degree representing a degree that power acquired by the power acquisition unit in a case that a predetermined reference speech signal is inputted into the power acquisition unit corresponds with predetermined reference power is higher than a predetermined reference correspondence degree, based on the probability distribution acquired by the probability distribution acquisition unit.

Abstract: A mobile device, such as a cellular telephone includes a voice interface that includes one part that may not be specific to a particular carrier, and a second part that provides an interface to services that are specific to a carrier or to service or information providers that are not necessarily available with all carriers. A voice command interface provides easy access to the carrier services. The set of carrier services is optionally extendible by the carrier.

Abstract: Methods and devices for wearable sound processing and voice operated control are provided. The method can include monitoring a sound pressure level between a first received sound and a second received sound, identifying a voicing level from a comparison of the sound pressure level, determining if a wearer is speaking based on the comparison and the voicing level, and transmitting a decision that the wearer's spoken voice is present to at least one among a cell phone, a media player, and a portable computing device. Other embodiments are disclosed.

Abstract: The present invention relates to means and methods of automated difference recognition between speech and music signals in voice communication systems, devices, telephones, and methods, and more specifically, to systems, devices, and methods that automate control when either speech or music is detected over communication links. The present invention provides a novel system and method for monitoring the audio signal, analyze selected audio signal components, compare the results of analysis with a pre-determined threshold value, and classify the audio signal either as speech or music.

Abstract: Disclosed an is an electronic equipment including: a command information storage section to store command pieces for controlling the electronic equipment each associated with a plurality of processing contents pertaining respectively to operation control in a plurality of operation states; a speech information obtainment section to obtain speech information; a command information extraction section to perform speech recognition of the obtained speech information so as to extract the corresponding command information stored in the command information storage section; a judgment section to judge an operation state of the electronic equipment when the command information is extracted; a control section to extract one of the plurality of pieces of control information associated with the extracted command information from the command information storage section based on the judged operation state of the electronic equipment so as to control the television receiver based on the extracted control information.

Abstract: The invention relates to a method, computer, computer program and computer program product for speech quality estimation. The method comprises the steps of: determining a coding distortion parameter (QCOD), a bandwidth related distortion parameter (BW) and a presentation level distortion parameter (PL) of a speech signal; extracting a first coefficient (?l) and a second coefficient (?2), the first coefficient and the second coefficient being dependent on the coding distortion parameter; and calculating a signal quality measure (Q), where the signal quality measure is QCOD+?1BW+?2PL using the signal quality measure in a quality estimation of the speech signal.

Abstract: An emotive advisory system for use by one or more occupants of an automotive vehicle includes a directional speaker array, and a computer. The computer is configured to determine an audio direction, and output data representing an avatar for visual display. The computer is further configured to output data representing a spoken statement for the avatar for audio play from the speaker array such that the audio from the speaker array is directed in the determined audio direction. A visual appearance of the avatar and the spoken statement for the avatar convey a simulated emotional state.

Abstract: Methods and devices for voice operated control are provided. The method can include 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: Methods and devices for voice operated control are provided. The method can include 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, or a spectral difference.

Abstract: A wireless headset system including a switching unit having a detection circuit coupled to a wireless adapter. The detection circuit provides a first output signal to the wireless adapter upon detection of a triggering event, such as activation of a manual switch or detection of voltage of an audio output signal from a mobile communications device. The first output signal causes the wireless adapter to enter into an active audio channel mode. The detection circuit also provides a second output signal to the wireless adapter when no triggering event is detected, and the second output signal causes the wireless adapter to enter into a standby mode. Placing at least the wireless adapter in a standby mode saves battery life of the system.

Abstract: Apparatus and methods for reducing misinterpretation of gesture-based input to portable electronic devices are described.

Abstract: An input signal is converted to a feature-space representation. The feature-space representation is projected onto a discriminant subspace using a linear discriminant analysis transform to enhance the separation of feature clusters. Dynamic programming is used to find global changes to derive optimal cluster boundaries. The cluster boundaries are used to identify the segments of the audio signal.

Abstract: A headset includes a wearable body, first and second earphones extending from the wearable body, controls for controlling an external communication/multimedia device wirelessly, a microphone for picking up vocal data from a user of the headset system and a signal processing unit. The signal processing unit includes circuitry for processing the vocal data into a distinctly audible vocal feedback signal, circuitry for enhancing the vocal feedback signal thereby producing an enhanced vocal feedback signal and circuitry for mixing the enhanced vocal feedback signal with audio signals originating from the external communication/multimedia device, thereby producing a mixed output signal and then sending the mixed output signal to the user via the earphones. The external communication/multimedia device comprises a vocal command application and the headset further comprises a vocal command control for sending vocal commands to the external communication/multimedia device and to the vocal command application.

Abstract: A gaming unit and producing a game-audio output of a varying volume level and a chat-voice output is connected to headphones adapted to be worn by a player of the unit and connected to the gaming unit for making the chat-voice output and the game-audio output audible to the player. A circuit or software associated with the headphones can increase the chat-voice volume level generally proportionately to the game-audio volume level.

Abstract: A device for booting a handheld apparatus by sound detection includes a base, a power-on device, a trigger switch, and an acoustic sensor. Upon the handheld apparatus being placed at the base to trigger the trigger switch, the trigger switch controls the power-on device to power on the handheld apparatus. After the handheld apparatus is powered on, the acoustic sensor detects a sound of the handheld apparatus and then controls a pressure head of the power-on device to move away. The device and its method for booting a handheld apparatus by sound detection come with the advantages of a simple and easy operation and a high efficiency.

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: 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 for controlling consumer exposure to audio dosage for a composite audio signal, the method comprising: sampling the composite audio signal; transforming the sampled audio signal, using a fast fourier transform algorithm, to produce a signal representative of the amplitude of component frequencies of the audio signal; comparing the transformed audio signal with a predefined impulse control threshold profile, representing the target maximum amplitude for each frequency component for the audio signal threshold profile to produce a configuring signal representative of the difference between the broadcast signal and the profile; using the configuring signal to automatically configure in real time a Finite Impulse Response (FIR) filter so that it attenuates the amplitude of the transformed audio signal in frequency bands centered on the frequencies at which the target threshold is exceeded; and outputting the attenuated audio signal for consumer exposure.

Abstract: Disclosed is a method of efficiently removing noise. The method includes: deciding a noise section by attenuating characteristics of a voice in a voice signal mixed with noise; determining the type of the noise in the decided noise section; and removing the noise from the noise-mixed voice signal by using noise information obtained through the determination. A clustering method or a similarity level measurement method is used to determine the type of the noise. Even in a voice signal that is mixed with various types of noise, noise can be precisely removed and thus distortion of a sound quality can be minimized.

Abstract: To provide an enhanced true-to-life atmosphere enjoyed in multipoint connecting, and reduce a calculation load at a multipoint connection unit, as well.

Abstract: A wireless headset system including a switching unit having a detection circuit coupled to a wireless adapter. The detection circuit provides a first output signal to the wireless adapter upon detection of a triggering event, such as activation of a manual switch or detection of voltage of an audio output signal from a mobile communications device. The first output signal causes the wireless adapter to enter into an active audio channel mode. The detection circuit also provides a second output signal to the wireless adapter when no triggering event is detected, and the second output signal causes the wireless adapter to enter into a standby mode. Placing at least the wireless adapter in a standby mode saves battery life of the system.

Abstract: A voice acquisition system for a vehicle includes an interior rearview mirror assembly attached at an inner portion of the windshield of a vehicle equipped with the interior rearview mirror assembly. The interior rearview mirror assembly includes at least two microphones for receiving audio signals within a cabin of the vehicle and generating an output indicative of the audio signals. A control is in the vehicle and is responsive to the output from the at least one microphone. The control at least partially distinguishes vocal signals from non-vocal signals present in the output. The at least two microphones provide sound capture for at least one of a hands free cell phone system, an audio recording system and a wireless communication system.

Abstract: A user interface device provides secure access to equipment. An audio reception device is adapted to receive an audio input comprising background noise and to receive an audio user code from a user. A user input device receives a manual input comprising a manual user code from the user. A control unit stores one or more target user codes and receives an audio input and determines a noise level of the background noise. An instruction message is generated to inform the user to enter the manual user code in response to the noise level of the background noise exceeding a predetermined threshold level. An output device provides the instruction message to the user. A control unit provides secure access to the equipment in response to at least one of the audio user code and the manual user code.