145results about How to "Easy to hear" patented technology

This invention is directed to providing voice feedback to a user of an electronic device. Because each electronic device display may include several speakable elements (i.e., elements for which voice feedback is provided), the elements may be ordered. To do so, the electronic device may associate a tier with the display of each speakable element. The electronic device may then provide voice feedback for displayed speakable elements based on the associated tier. To reduce the complexity in designing the voice feedback system, the voice feedback features may be integrated in a Model View Controller (MVC) design used for displaying content to a user. For example, the model and view of the MVC design may include additional variables associated with speakable properties. The electronic device may receive audio files for each speakable element using any suitable approach, including for example by providing a host device with a list of speakable elements and directing a text to speech engine of the host device to generate and provide the audio files.

By monitoring the wind noise in a location in which a cellular telephone is operating and by applying noise reduction and / or cancellation protocols at the appropriate time via analog and / or digital signal processing, it is possible to significantly reduce wind noise entering into a communication system.

A clear, high quality voice signal with a high signal-to-noise ratio is achieved by use of an adaptive noise reduction scheme with two microphones in close proximity. The method includes the use of two omini directional microphones in a highly directional mode, and then applying an adaptive noise cancellation algorithm to reduce the noise.

Various embodiments of the present invention enable a system and method for reducing or entirely canceling background or environmental noise from a voice transmission from a communications device. A communications device, such as a mobile telephone, is configured with an environmental noise compensation, correction, or counterbalanced (correction) signal generator that is connected between at least one microphone and a continuous time quadrant multiplication. Optional discrete time or digital signal processing may be applied. The original output of the at least one microphone and a compensation, correction, or counterbalanced or counter-balancing signal generated by the environmental noise compensation, correction, or counterbalanced signal generator are mixed or otherwise combined together after being received by the antenna to the receiver.

Disclosed is a sound output device for electric vehicle including a sound producing unit 11 for producing a sound, an information acquiring unit 12 for acquiring state information about the state of a vehicle or surroundings information about surroundings of the vehicle, and a control unit 13 for controlling a sound volume and an output direction of the sound produced by the sound producing unit 11 according to the vehicle state information or the surroundings information acquired by the information acquiring unit 12 to output the sound to outside the vehicle. Therefore, the sound output device for electric vehicle makes it easy for pedestrians and passengers each riding on a vehicle, such as a light vehicle or a motorbike, to hear the sound produced thereby, or can prevent neighboring residents from being put into an unpreferable situation.

The invention provides a volume adjusting method comprising the following steps of: A. determining the set difference value of the volume of an audio signal output by an audio playing device and the detected volume of environmental background noise during setting; B. detecting the volume of the current environmental background noise and the frequency of the audio signal; C. adjusting the current volume of the audio playing device for outputting the audio signal according to the summation of the detected volume of the current environmental background noise and the set difference value ; D. determining each grain value for adjusting each frequency of the audio signal when the adjusted volume is judged to exceed the preset volume range; and E. adjusting the frequency of the audio frequency playing device for outputting the audio signal according to the determined grain value. The algorithm is simpler and can be used for adjusting the volume more quickly. The invention also provides a volume adjusting device, which can be used for adjusting the volume more quickly.

A noise reduction system and method suitable for hands-free communication comprises a first microphone located in a wireless or cordless headset and a second microphone located within a separate communication device. The first microphone may be placed within a headset attached to or near the ear of the user while the second microphone may be located further away from the user's mouth. The first microphone collects the intended voice signal and ambient background or environmental noise while the second microphone primarily collects ambient noise rather than a voice signal. The inputs to the two microphones are synchronized and the ambient noise of the second microphone is used to compensate or remove noise from the first microphone, such as by subtracting the noise from the second microphone from the input of the first microphone. Noise reduction or noise cancellation is achieved through continuous time quadrant modulation, environmental noise counterbalancing, and / or through other means.

A noise reduction system and method suitable for hands-free communication comprises a first microphone located in a wireless or cordless headset and a second microphone located within a separate communication device. The first microphone may be placed within a headset attached to or near the ear of the user while the second microphone may be located further away from the user's mouth. The first microphone collects the intended voice signal and ambient background or environmental noise while the second microphone primarily collects ambient noise rather than a voice signal. The inputs to the two microphones are synchronized and the ambient noise of the second microphone is used to compensate or remove noise from the first microphone, such as by subtracting the noise from the second microphone from the input of the first microphone. Noise reduction or noise cancellation is achieved through continuous time quadrant modulation, environmental noise counterbalancing, and / or through other means.

The present invention relates to hearing aid training systems (100). More particularly, the present invention relates to the simulation of a hearing aid environment (108) prior to a user's (105) purchase of a hearing aid. To create the simulated environment, the user's hearing profile (111) is collected from all prior hearing tests. Prior hearing tests include information on all aspects of the user's hearing, such as frequency and speech intelligibility. The software program (126) of this invention, and the audiologist using the software program (126), analyzes the user's hearing profile and creates a simulation that demonstrates to the user how he or she would hear with a hearing aid. Furthermore, this invention provides a way to make additional adjustments to the hearing aid's DSP data based upon user preferences prior to ordering the individual customized hearing aid.

An audio player includes a memory storing one or more audio data files and at least one personalized audio profile comprising for each ear a map of amplitude audio frequency profile. A file selector permits a user to select one or more of the stored audio data files and at least one of the personalized profiles. A data processor accesses the selected files and profiles from the memory and processes the selected files with the profile to generate a processed audio signal for each ear. An actuator reproduces the audio information stored in the processed audio signal as sound for each ear. Personalized profiles are produced by monitoring and storing the responses of users to audible signals having different frequencies.

The present universal wearable computing device relates to a hearing assistance system, device, method, and apparatus that provide a discreet approach to user hearing assistance, without relying on a conventional hearing aid. The hearing assistance system and the requisite electronics may be incorporated into frames that also function as eyeglasses with earphone(s) that may be connected to the frame to assist user hearing. An earphone may be configured with minimal electronics, such that a power source enable sound transmissions to the ear, is provided by a connection to the frame of the eyeglasses. In another example, the earphone is configured without any electronics and sound is transmitted to the user / listener's ear(s) via a psychoacoustic system. The sound quality of the transmissions to the earphones may be optimized using a tuning / equalizer application operating from a computing device, such as an app on a mobile device. The tuning / equalizer application can be used by the user / listener to optimize volume input levels to the earphone(s). The hearing assistance system may also protect from damaging environmental noise, provide heightened hearing capabilities, collect and process physiological and physical measurements, and provide communication capabilities for interacting with other computing devices.

A system of communication for providing instruction, information, and verbal commands between a teacher and a student. An exemplary embodiment of the invention is particularly suited for use between a sports coach and the players on his or her team. The system of the present invention is preferably comprised of a microphone for the coach, a transmitter, at least one receiver, and at least one speaker for at least one player.

A system and method for reducing or entirely canceling background or environmental noise from a voice transmission from a communications device. A communications device, such as a cellular telephone or cordless telephone, is configured with an environmental noise counterbalanced (correction) signal generator that is connected between a microphone and a continuous time quadrant multiplication. The original output of the microphone and a counterbalanced (correction) signal generated by the environmental noise counterbalanced (correction) signal generator are mixed or combined together prior to being passed to a transmitter. In one embodiment a discrete time unit (discrete time processing) block with or without signal processing is provided between the microphone and the continuous time quadrant multiplication to help synchronize the timing of the signals to be mixed. In another embodiment a second microphone is employed to detect environmental noise.

VoIP phones according to the present invention include a microphone, which may be internal or external, and allow the user to communicate unobtrusively, check voice mail and conduct other activities in an environment which can be noisy in general and extremely noisy sometimes. Speech recognition functionally may also be used to generate and send touch tone or DTMF tones such as in response to call trees or voice recognition functionality used by airlines, credit card companies, voice mail systems, and other applications. A system and method of audio processing which provides enhanced speech recognition is provided. Audio input is received at the microphone which is processed by adaptive noise cancellation to generate an enhanced audio signal. The operation of the speech recognition engine and the adaptive noise canceller may be advantageously controlled based on Voice Activity Detection (VAD).

A system and method for reducing or entirely canceling background or environmental noise from a voice transmission from a communications device. A communications device, such as a voice over internet or voice over packets mobile or wire lined telephone or cordless telephone, is configured with an environmental noise counterbalanced (correction) signal generator that is connected between a microphone and a continuous time quadrant multiplication. The original output of the microphone and a counterbalanced (correction) signal generated by the environmental noise counterbalanced (correction) signal generator are mixed together prior to being passed to a transmitter. In one embodiment a discrete time unit (discrete time processing) block with or without signal processing is provided between the microphone and the continuous time quadrant multiplication to help synchronize the timing of the signals to be mixed. In another embodiment a second microphone is employed to detect environmental noise.

A clear, high quality voice signal with a high signal-to-noise ratio is achieved by use of an adaptive noise reduction scheme with two microphones in close proximity. The method includes the use of two omini directional microphones in a highly directional mode, and then applying an adaptive noise cancellation algorithm to reduce the noise.

Provided are a hearing aid, signal processing method, and program that enable TV sound to be made easier for a hearing aid user to hear when wishing to watch TV, and a person's voice to be made easier to hear when wishing to converse with that person. A hearing aid (100) is provided with an other-person's speech detection section (150) that detects speech of a speaker other than the wearer using detected sound source direction information, an own-speech detection result, and a TV sound detection result, and a per-sound-source frequency calculation section (160) that calculates the frequency of each sound source using an own-speech detection result, TV sound detection result, other-speaker's speech detection result, and sound source direction information. A scene determination section (170) determines a scene to be a “conversation scene,”“TV viewing scene,” and “‘TV viewing while . . . ’ scene” using sound source direction information and the per-sound-source frequency, and an output sound control section (180) controls hearing of the hearing aid (100) according to a determined scene.

A sound input device includes a differential microphone, configured to receive sound including noise, and generate a first signal in accordance with the sound; a detector, configured to detect the noise, and generate a second signal in accordance with the detected noise; and a controller, configured to control at least one of suppression of high-frequency components of the first signal and changing of a frequency band to be suppressed of the first signal based on the second signal.

While the amplification factor of an external amplifier is increased to raise the minimum volume level, synthesized speech of text data is generated as PCM data. When the PCM data is output for each frame (63 ms), the voltage of a battery power supply is detected and different ATT values (attenuation ratios) are set in accordance with threshold values (Level 1˜4) corresponding to the detected voltage. The lower the power supply voltage, the larger the attenuation ratios. The PCM data is attenuated by such an attenuation ratio to reduce a load to the battery power supply. Even though the battery power supply is exhausted to some extent, no power-down due to an instantaneous increase in power load occurs, with the result that a high-level part of a series of speech data items is output at an adequate volume level that makes a low-level part thereof easy to hear.

VoIP phones according to the present invention include a microphone, which may be internal or external, and allow the user to communicate unobtrusively, check voice mail and conduct other activities in an environment which can be noisy in general and extremely noisy sometimes. Speech recognition functionally may also be used to generate and send touch tone or DTMF tones such as in response to call trees or voice recognition functionality used by airlines, credit card companies, voice mail systems, and other applications. A system and method of audio processing which provides enhanced speech recognition is provided. Audio input is received at the microphone which is processed by adaptive noise cancellation to generate an enhanced audio signal. The operation of the speech recognition engine and the adaptive noise canceller may be advantageously controlled based on Voice Activity Detection (VAD).

There is provided a surround device which processes stereo signals decoded by a decoder of a multi-channel surround method comprises boost means which boosts middle and low frequency components of a center channel signal in the stereo signals. There is also provided a surround device which processes stereo signals decoded by a decoder of a Dolby Pro Logic method comprises boost means which boosts high frequency components of a surround channel signal in the stereo signals. The boost means is composed of a level detector for detecting an output level of the channel signal, an operation portion for calculating and determining an amount of boost based on a result of the level detector, and a signal boost portion for boosting the channel signal based on a result of the operation portion.

The invention relates to the field of computers and wireless communication, in particular to a mobile terminal-oriented information service providing system and a mobile terminal-oriented information service providing method. The method comprises the following steps that: a centre server of news information is established, wherein the news of the centre server is collected and classified by associated personnel of the center through a maintenance system and is placed on a platform by the associated personnel and types of the news can be dynamically added or deleted during different periods; a user controls the mobile terminal to request downloading the classified news information to the center server through a cellular mobile communication network and downloading local news content; and the user controls the mobile terminal to display the news on a mobile terminal screen and can control whether to call a TTS voice module to synthesize the news into voice and play the voice through an amplifier or not. By the technical scheme of the invention, the user can acquire the latest information at any time and at any place and can listen to news easily under the condition that the user is incapable of reading or is inconvenient to read texts.

A microphone unit includes: a partition member, including a vibration film configured to be vibrated by sound; a housing, defining an inner space, and formed with a first opening and a second opening, the inner space being divided into a first space and a second space by the partition member, the first space adapted to be communicated with an outer space via the first opening, and the second space adapted to be communicated with the outer space via the second opening; and a shutter, configured to close one of the first and second openings; and an amplifier, configured to: amplify an electric signal at a first gain when the shutter closes the one of the first and second openings; and amplify the electric signal at a second gain that is larger than the first gain when first and second openings are opened.

The invention discloses a voice playing method. The method comprises the steps of determining whether a playing frequency of a voice message in a preset time segment exceeds a preset threshold or not when a first clicking operation of a user on the voice message is detected; if the playing frequency of the voice message in the preset time segment exceeds the preset threshold, determining target playing volume and a target playing speed, wherein the target playing volume is higher than playing volume of a currently played voice message, and the target playing speed is smaller than a playing speed of the currently played voice message; and playing the voice message according to the target playing volume and the target playing speed. An embodiment of the invention furthermore provides a terminal device. With the adoption of embodiments of the method and the terminal device, the listening efficiency of the voice message can be improved.

A system and method for reducing or entirely canceling background or ambient noise from a voice transmission from a communications device. A communications device, such as a mobile telephone, is configured with an ambient noise compensation signal generator that is connected between a microphone and a mixer. The original output of the microphone and a compensation signal generated by the ambient noise compensation signal generator are mixed together prior to being passed to a transmitter. In one embodiment a buffer is provided between the microphone and the mixer to help synchronize the timing of the signals to be mixed. In another embodiment a second microphone is employed to detect ambient noise.

An orthopedic driver is provided that includes an elongate tubular body and an indicator assembly. The indicator assembly has a housing that has at least one opening and a visual indicator disposed in the at least one opening. The visual indicator is configured to be coupled with a guide pin such that lateral movement of the elongate tubular body relative to the guide pin causes the visual indicator to move between a flush or recessed configuration within the housing and an extended configuration. In the extended configuration, the visual indicator extends outward of an outer surface of the housing. The flush or recessed configuration corresponds to alignment of the guide pin relative to the elongate tubular body and the extended configuration corresponds to misalignment of the guide pin relative to the elongate tubular body.

A headset device with active noise reduction which can switch to a hearing aid mode in which outside sound is picked up and directed into the ear or amplified to make it easy to hear comprises a housing that is mountable near or into a human ear, a speaker unit equipped within the housing and capable of emitting a sound signal, one or a plurality of microphone units for collecting ambient sound; and a signal processor electrically connected to the microphone unit(s), wherein the signal processor selectively operates, either in a hearing aid mode in which ambient sound coming in from the outside through the microphone unit(s) is emitted, or in a noise reduction mode in which ambient sound coming in from the outside is inverted and the inverted ambient sound is emitted through the speaker unit to reduce that ambient sound.

By monitoring the wind noise in a location in which a cellular telephone is operating and by applying noise reduction and / or cancellation protocols at the appropriate time via analog and / or digital signal processing, it is possible to significantly reduce wind noise entering into a communication system.

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.