2284 results about "Environmental noise" patented technology

The present invention relates to a non-invasive device for measuring physiological processes. More particularly, it concerns a device that can be applied externally to the body of an animal or human to detect and quantify displacement, force, motion, vibration and acoustic effects resulting from internal biological functions. Specifically, an inexpensive device is disclosed that is compact, light, portable and comfortable, and operates satisfactorily even with imprecise location on the body, ambient noise, motion and light.

The present invention provides a speech recognition system combined with one or more alternate input modalities to ensure efficient and accurate text input. The speech recognition system achieves less than perfect accuracy due to limited processing power, environmental noise, and / or natural variations in speaking style. The alternate input modalities use disambiguation or recognition engines to compensate for reduced keyboards, sloppy input, and / or natural variations in writing style. The ambiguity remaining in the speech recognition process is mostly orthogonal to the ambiguity inherent in the alternate input modality, such that the combination of the two modalities resolves the recognition errors efficiently and accurately. The invention is especially well suited for mobile devices with limited space for keyboards or touch-screen input.

A system is disclosed for detecting and calculating the level of ambient and / or environmental noise, such as electromagnetic interference generated by electric power lines, ambient lights, light dimmers, television or computer displays, power supplies or transformers, and medical equipment. In some embodiments, the system performs frequency analysis on the interference signal detected by light photodetectors and determines the power of the interference signal concentrated in the analyzed frequency bands. The worst-case interference level can be determined by selecting the maximum from the computed power values. In some embodiments, the determined interference signal power can be compared with the noise tolerance of a patient monitoring system configured to reliably and non-invasively detect physiological parameters of a user. The results of the comparison can be presented to the user audio-visually. In some embodiments, the system can be used to perform spot check measurements of electromagnetic interference.

The present invention provides a speech recognition system combined with one or more alternate input modalities to ensure efficient and accurate text input. The speech recognition system achieves less than perfect accuracy due to limited processing power, environmental noise, and / or natural variations in speaking style. The alternate input modalities use disambiguation or recognition engines to compensate for reduced keyboards, sloppy input, and / or natural variations in writing style. The ambiguity remaining in the speech recognition process is mostly orthogonal to the ambiguity inherent in the alternate input modality, such that the combination of the two modalities resolves the recognition errors efficiently and accurately. The invention is especially well suited for mobile devices with limited space for keyboards or touch-screen input.

The present invention provides a speech recognition system combined with one or more alternate input modalities to ensure efficient and accurate text input. The speech recognition system achieves less than perfect accuracy due to limited processing power, environmental noise, and / or natural variations in speaking style. The alternate input modalities use disambiguation or recognition engines to compensate for reduced keyboards, sloppy input, and / or natural variations in writing style. The ambiguity remaining in the speech recognition process is mostly orthogonal to the ambiguity inherent in the alternate input modality, such that the combination of the two modalities resolves the recognition errors efficiently and accurately. The invention is especially well suited for mobile devices with limited space for keyboards or touch-screen input.

The ability to combine multiple audio signals captured from the microphones in a microphone array is frequently used in beamforming systems. Typically, beamforming involves processing the output audio signals of the microphone array in such a way as to make the microphone array act as a highly directional microphone. In other words, beamforming provides a “listening beam” which points to a particular sound source while often filtering out other sounds. A “generic beamformer,” as described herein automatically designs a set of beams (i.e., beamforming) that cover a desired angular space range within a prescribed search area. Beam design is a function of microphone geometry and operational characteristics, and also of noise models of the environment around the microphone array. One advantage of the generic beamformer is that it is applicable to any microphone array geometry and microphone type.

Ambient noise sampled by a mobile device from a local environment is used to automatically trigger actions associated with content currently playing on the mobile device. In some implementations, subtitles or closed captions associated with the currently playing content are automatically invoked and displayed on a user interface based on a level of ambient noise. In some implementations, audio associated with the currently playing content is adjusted or muted. Actions can be automatically triggered based on a comparison of the sampled ambient noise, or an audio fingerprint of the sampled ambient noise, with reference data, such as a reference volume level or a reference audio fingerprint. In some implementations, a reference volume level can be learned on the mobile device based on ambient noise samples.

Method and apparatus for audio intelligibility enhancement and computing apparatus are provided. The method includes the following steps. Environment noise is detected by performing voice activity detection according to a detected audio signal from at least a microphone of a computing device. Noise information is obtained according to the detected environment noise and a first audio signal. A second audio signal is outputted by boosting the first audio signal under an adjustable headroom by the computing device according to the noise information and the first audio signal.

A connector having a conductive member is provided, wherein the connector comprises a connector body capable of sealing and securing a coaxial cable, and further wherein the conductive member, such as an O-ring, physically seals the connector, electrically couples the connector and the coaxial cable, facilitates grounding through the connector, and renders an electromagnetic shield preventing ingress of unwanted environmental noise.

An optical probe, which is particularly suited to reduce noise in measurements taken on an easily compressible material, such as a finger, a toe, a forehead, an earlobe, or a lip, measures characteristics of the material. A neonatal and adult disposable embodiment of the probe include adhesive coated surfaces to securely affix the probe onto the patient. In addition, the surface of the probe is specially constructed to reduce the effect of ambient noise.

A sensor panel device that can generate and use a stimulus signal having multiple different waveforms for detecting events on or near the sensor panel is disclosed. Among other things, such a stimulus signal can be used to reject environmental noise present in the device. In some embodiments, the stimulus signal has multiple waveforms having different frequencies. Logic circuitry can generate representative values from output of the different waveforms applied to one or more sensing nodes in the sensor panel device. From the representative values, a final value can be generated that represents whether an event occurred at or near the one or more sensing nodes.

A cabin communication system for improving clarity of a voice spoken within an interior cabin having ambient noise includes an adaptive speech enhancement filter for receiving an audio signal that includes a first component indicative of the spoken voice, a second component indicative of a feedback echo of the spoken voice and a third component indicative of the ambient noise, the speech enhancement filter filtering the audio signal by removing the third component to provide a filtered audio signal, the speech enhancement filter adapting to the audio signal at a first adaptation rate, and an adaptive acoustic echo cancellation system for receiving the filtered audio signal and removing the second component in the filtered audio signal to provide an echo-cancelled audio signal, the echo cancellation signal adapting to the filtered audio signal at a second adaption rate, wherein the first adaptation rate and the second adaptation rate are different from each other so that the speech enhancement filter does not adapt in response to operation of the echo-cancellation system and the echo-cancellation system does not adapt in response to operation of the speech enhancement filter.

A method and apparatus are provided for identifying a noise environment for a frame of an input signal based on at least one feature for that frame. Under one embodiment, the noise environment is identified by determining the probability of each of a set of possible noise environments. For some embodiments, the probabilities of the noise environments for past frames are included in the identification of an environment for a current frame. In one particular embodiment, a count is generated for each environment that indicates the number of past frames for which the environment was the most probable environment. The environment with the highest count is then selected as the environment for the current frame.

A method and system for enhancing a target sound signal from multiple sound signals is provided. An array of an arbitrary number of sound sensors positioned in an arbitrary configuration receives the sound signals from multiple disparate sources. The sound signals comprise the target sound signal from a target sound source, and ambient noise signals. A sound source localization unit, an adaptive beamforming unit, and a noise reduction unit are in operative communication with the array of sound sensors. The sound source localization unit estimates a spatial location of the target sound signal from the received sound signals. The adaptive beamforming unit performs adaptive beamforming by steering a directivity pattern of the array of sound sensors in a direction of the spatial location of the target sound signal, thereby enhancing the target sound signal and partially suppressing the ambient noise signals, which are further suppressed by the noise reduction unit.

Embodiments of the invention relate to a feedback mechanism that informs a user of a communication device to adjust the volume, pitch, tone or other characteristic of his voice so as to compensate for noise in the surrounding environment. The feedback mechanism includes feedback circuitry that analyzes audio signals from the microphone and preferably from one or more additional environmental noise sensors. From the analysis, the feedback circuitry determines characteristics of the user's voice and characteristics of the environmental noise, and provides an analysis of how the user might modify his voice to best compensate for environmental noise. This analysis results in an indication to the user, such as through a vibration, a sound, or graphical indication on the device, which tells the user whether and to what extent the user should adjust a characteristic of his voice to best overcome such environmental noise.

Intelligibility of a human voice projected by a loudspeaker in an environment of high ambient noise is enhanced by processing a voice signal in accordance with the frequency response characteristics of the human hearing system. Intelligibility of the human voice is derived largely from the pattern of frequency distribution of voice sounds, such as formants, as perceived by the human hearing system. Intelligibility of speech in a voice signal is enhanced by filtering and expanding the voice signal with a transfer function that approximates an inverse of equal loudness contours for tones in a frontal sound field for humans of average hearing acuity.

A connector having a conductively coated member is provided, wherein the connector comprises a connector body capable of sealing and securing a coaxial cable, and further wherein the conductively coated member, such as an O-ring, physically seals the connector, electrically couples the connector and the coaxial cable, facilitates grounding through the connector, and renders an electromagnetic shield preventing ingress of unwanted environmental noise.

Natural-quality synthetic noise will replace background acoustic noise during speech gaps and will achieve a better representation of the excitation signal in a noise-synthesis model by classifying the type of acoustic environment noise into one or more of a plurality of noise classes. The noise class information is used to synthesize background noise that sounds similar to the actual background noise during speech transmission. In some embodiments, the noise class information is derived by the transmitter and transmitted to the receiver which selects corresponding excitation vectors and filters them using a synthesis filter to construct the synthetic noise. In other embodiments, the receiver itself classifies the background noise present in hangover frames and uses the class information as before to generate the synthetic noise. The improvement in the quality of synthesized noise during speech gaps helps to preserve noise continuity between talk spurts and speech pauses, and enhances the perceived quality of a conversation.

A mobile phone or other personal communication device includes resources applying measures of an individual's hearing profile, personal choice profile, and induced hearing loss profile, separately or in combination, to build the basis of sound enhancement. A personal communication device thus comprises a transmitter / receiver coupled to a communication medium for transmitted receiving audio signals, control circuitry that controls transmission, reception and processing of call and audio signals, a speaker, and a microphone. The control circuitry includes logic applying one or more of a hearing profile of the user, a user preference related hearing, and environmental noise factors in processing the audio signals. The control circuitry may includes instruction memory and an instruction execution processor such as a digital signal processor.

A cabin communication system for improving clarity of a voice spoken within an interior cabin having ambient noise includes a microphone for receiving the spoken voice and the ambient noise and for converting the spoken voice and the ambient noise into an audio signal, the audio signal having a first component corresponding to the spoken voice and a second component corresponding to the ambient noise, a speech enhancement filter for removing the second component from the audio signal to provide a filtered audio signal, the speech enhancement filter removing the second component by processing the audio signal by a method taking into account elements of psycho-acoustics of a human ear, and a loudspeaker for outputting a clarified voice in response to the filtered audio signal.

A remote control system includes a remote control transmitter that transmit a signal and a receiver than can receive the signal from the transmitter. A control device performs a function having a first and a second option. The control device responds to the signal received by the receiver and performs the first option if the distance between the remote control transmitter and the receiver is greater than a particular distance and performs a function based on the ambient noise level.

Systems and methods are described that automatically adjust a value of a parameter relating to the delivery of media content, such as audio content or image content, based on both environmental conditions and on automatically-learned user preference data. For example, a first embodiment adjusts a volume setting used to control the delivery of an audio signal based both on environmental noise conditions and upon automatically-learned user preference information, wherein the user preference information is derived by monitoring user-implemented adjustments to the volume setting after application of an automatic adjustment thereto. As another example, a second embodiment adjusts a brightness setting used to control the brightness of a display used for rendering images based both on an ambient light level and upon automatically-learned user preference information, wherein the user preference information is derived by monitoring user-implemented adjustments to the brightness setting after application of an automatic adjustment thereto.

Provided is a detector system for detecting a plurality of items each having at least one RFID tag attached thereto. The detector system may include a presence detector and / or a message detector. The presence detector may include a signal reader which receives modulated carrier frequencies from the RFID tags during a predetermined response time period for comparison to a reference level. The signal reader may be configured as a power integrator or a constellation discriminator. The power integrator sums the power contained in signals transmitted by the RFID tags by integration for comparison to an environmental noise level measurement taken during a quiet period. For the constellation discriminator, bits contained in the modulating message are compared to the bit pattern of a predetermined reference constellation. The detector system may be configured as a 3-state detector which combines the presence detector with a message detector that indicates the identification code of the RFID tags.

A speech recognizer operating in both ambient noise (additive distortion) and microphone changes (convolutive distortion) is provided. For each utterance to be recognized the recognizer system adapts HMM mean vectors with noise estimates calculated from pre-utterance pause and a channel estimate calculated using an Estimation Maximization algorithm from previous utterances.

Music information is displayed during playback of multimedia content performed by a multimedia playback device. The multimedia content comprises audio and video. To obtain the music information, audio of the multimedia content is retrieved, environmental noise is filtered out of the audio to generate filtered audio, a rhythm of the filtered audio is obtained, and the music information is generated based on the rhythm of the filtered audio. The music information is displayed simultaneously during playback of the multimedia content.

A first device obtains first audio from one or more microphones. The first device generates a first audio signal from the first audio. The first device analyzes the first audio signal to determine noise associated with the first audio. The first device receives a second audio signal from a second device, and processes the second audio signal based at least in part on the determined noise by identifying one or more frequencies of the second audio signal that are between 1-2 Kilohertz. The first device then outputs a modified second audio signal to a speaker.

An audio correcting apparatus includes a speaker provided on a television apparatus, a microphone provided on a remote controller, an identifying unit which identifies an acoustic characteristic from the speaker to the microphone, and an acoustic characteristic setting unit having the acoustic characteristic. A signal obtained by allowing an audio signal input to the speaker to pass through the acoustic characteristic setting unit, and a signal representing ambient noise are input to an audio-correcting filter and a loudness-compensation-gain calculating unit. Based on both signals, the sound pressure level of sound output from the speaker is corrected so that the sound output from the speaker is clearly heard when reaching the user without being affected by the ambient noise.

A light element includes a translucent shell and a base. The base supports a plurality of LEDs positioned so as to emit a light show through the shell. The device also includes a compartment for receiving and securing a placeable volatile active cartridge and a heater for enabling the device to effectively emit an active ingredient from the cartridge when the cartridge is secured in the compartment. A white light source is provided in the shell as a source of illumination along with the LEDs The device can be used as a white light source, for displaying a colored light show, for displaying colored light schemes and for volatile active emission The devise may include an acoustic transducer for producing light and sound show. The devise may also include a light sensor, a motion detector and / or a microphone or acoustic receiver thereby enabling the light and sound show to be dependent upon ambient light intensity, motion detection and or / ambient noise

A method and apparatus for managing communication between participants in a virtual environment enables the participants to elect to automatically connect with each other via a preferred communication mechanism, manually connect with each other, or choose not to connect with each other. The connection may be point-to-point between two participants or may include multiple participants. Establishment of a connection, or the ability to establish a connection, may be based on the proximity of the participants, or the Avatars representing the participants, in the virtual environment. Once the connection is established, the connection may be managed so that the connection is maintained while the Avatars remain proximate each other and is automatically severed once the Avatars move away from each other. Environmental noise and other audio aspects may be included in the connection to simulate a real-world conversation.

Systems and methods for ambient noise compensation are disclosed. One example of a system includes a variable amplifier, a source sound processor, an area sound processor, and an adjustment circuit. The variable amplifier adjusts an audio input signal to generate an audio output signal with an appropriate level so that the audio output signal is audible over noise in a listening area. The source sound processor and the area sound processor may split the audio output signal and a monitoring signal into frequency bands, and may compare these signals band-by band to find differences that represent time-varying noise in the monitoring signal. These differences may be modified to account for the acoustic response of the listening area and for constant-level background noise in the listening area. The adjustment circuit controls the variable amplifier in response to these differences.