104 results about "Audio filter" patented technology

Disclosed herein are an apparatus and method for providing a haptic effect using a sound effect. The apparatus includes an audio filter storage unit, an acquisition unit, an analysis unit, a message configuration unit, and a haptic output unit. The audio filter storage unit stores a plurality of adaptive audio filters. The acquisition unit obtains sound effects output by an electronic device in response to an application or a user input event. The analysis unit analyzes the frequency components of each of the sound effects. The message configuration unit detects at least one of the adaptive audio filters from the audio filter storage unit, and generates a haptic output message, corresponding to the sound effect. The haptic output unit outputs a haptic effect based on the haptic output message. The adaptive audio filter dynamically varies depending on the application or the user input event.

A communications system may include at least one mobile wireless communications device, and a wireless communications network for sending text messages thereto. More particularly, the at least one mobile wireless communications device may include a wireless transceiver and a controller cooperating therewith for receiving text messages. The controller may be switchable between a normal message mode and an audio message mode. Also, the at least one mobile wireless communications device may further include a user interface device connected to the controller for receiving at least one audio mode filter parameter from a user, and an audio output connected to the controller. The controller, when in the audio message mode, may select received text messages based upon the at least one audio mode filter parameter, and output audio messages including speech generated from the selected text messages via the audio output.

A computerized notetaking system that records audio and links notes to the audio and has enhancements such as: Always-on audio recording and external timestamp button that work even when the system is turned off. A “next-topic” command that prepares and timestamps a new paragraph while allowing the user to complete the current paragraph. Commands for creating callouts and sidebar boxes in the user notes. A choice of audio filters. Speech recognition for searching through and navigating an audio recording. Speech recognition accuracy enhancement based upon typed notes. Playback, including lockstep playback, that prefers starting and stopping at word boundaries without overlap when possible. A self-adjusting preplay parameter. A “repeat slower” command that replays a few seconds of audio and then resumes at normal speed. Integrated background audio that plays music or white noise when notes-related audio is not being played. Several other enhancements are disclosed.

This invention discloses a synchronous play control method for supporting multi-source flow media, in which, a device realizing said method includes a separator, a decoder set, a multi-source flow media module, a multi-source video flow combination module, an OSD and an audio filter lens, in which, the separator separates the video and audio data in the multiple local media files or the multiple flow media, the decoder set calls related decoders to decode them to be sent to the multi-source flow media synchronous module, which applies a system of multi-size hierarchical synchronous control to control the multi-route media flow synchronously among and in the media objects, the multi-source video combination module combines multiple videos to one route, the OSD module laminates the combined data, volume, the current play time or caption information to output the video, the audio filter lens carries out formation transformation and linear overlap to the multiple audios then to output the audio.

A device used with a conventional television set that permits both hearing impaired and non hearing impaired individuals to view television audio voice signals in a text format on the television screen without the requirement of a closed / open caption broadcast signal or a closed / open caption enabled television. The device also can utilize an FM audio input, an audio line input, and a microphone or impedance input for conversion into a text format. The system may use on-board or remote displays, wireless or wired, for also providing the text format representative of human speech. The system includes audio filters for filtering a television audio signal and filtering out human speech audio signals for processing and conversion by a speech-to-text converter.

An arc fault detector configured to process audio frequencies associated with an arc fault condition to detect the occurrence of an arc fault condition. In one embodiment, the arc fault detector includes an alternating current (AC) line filter for receiving an AC line signal having a line frequency portion and an audio frequency portion in the electromagnetic spectrum associated with at least one arc condition. The AC line filter removes the line frequency portion and passes the audio frequency portion to an audio frequency filter which processes the audio frequency portion for an arc fault signature to detect for the occurrence of an arc fault condition.

There is disclosed a video playback device for playing back a video program in fast forward mode on a display screen coupled to the video playback device. The video playback device contains an apparatus for selectively playing back portions of an audio signal associated with the video program during fast forward mode. The apparatus comprises: 1) an audio filtering processor for identifying a plurality of audio sub-segments in the audio signal and determining an importance factor associated with each of the plurality of audio sub-segments; and 2) a video processor coupled to and controlled by the audio filtering processor, wherein the audio filtering processor transmits selected ones of the plurality of audio sub-segments to the video processor and causes the video processor to play the selected audio sub-segments concurrently with a corresponding portion of a video signal from the video program during fast forward mode.

An audio filter consists of two substantially identical stages with different purposes. The first stage (301) whitens detected noise, while preserving speech or other audible information in an undistorted manner. The second stage (303) effectively eliminates the residual white noise. Each stage, in one embodiment, includes a Mel domain based error minimization stage (108). A two stage Mel-frequency domain Wiener filter (300) is designed for each speech time frame in the Mel-frequency domain, instead of linear frequency domain. Each Mel domain based error minimization stage (108) minimizes the perceptual distortion and reduces the computation requirement to provide suitably filtered audible information.

Systems and methods of processing audio signals are described. The audio signals comprise information about spatial position of a sound source relative to a listener. At least one audio filter generates two filtered signals for each of audio signal. The two filtered signals are mixed with other filtered signals from other audio signals to create a right output audio channel and a left audio output channel, such that the spatial position of the sound source is perceptible from the right and left audio output channels.

A system and method for acoustic echo cancellation is provided. Embodiments may include receiving, at one or more microphones, an audio reference signal from an audio speaker. Embodiments may also include filtering the audio reference signal using one or more adaptive audio filters. Embodiments may further include analyzing a level of signal energy of the audio reference signal with regard to time, frequency and audio channel to identify at least one maximum error contribution point. Embodiments may also include updating the one or more adaptive audio filters based upon, at least in part, the analyzed audio reference signal.

A noise-suppression assembly of a mechanical drive system having a rotational frequency includes an audio filter unit configured to receive a first audio signal and a timing signal of the mechanical drive system. The audio filter unit generates a noise-cancellation signal based on a frequency of the timing signal to suppress a noise generated by the mechanical drive system and to apply the noise-cancellation signal to the first audio signal to produce a filtered first audio signal. The frequency of the timing signal is based on the rotational frequency of the mechanical drive system.

Systems and methods of processing audio signals are described. The audio signals comprise information about spatial position of a sound source relative to a listener. At least one audio filter generates two filtered signals for each of audio signal. The two filtered signals are mixed with other filtered signals from other audio signals to create a right output audio channel and a left audio output channel, such that the spatial position of the sound source is perceptible from the right and left audio output channels.

The present invention relates to baby monitoring system and method, and is especially one baby monitoring system and method utilizing baby's cry frequency and showing baby's state in image display equipment. The present invention includes baby's state detecting device and image display equipment. The baby's state detecting device consists of controller to control the operation of the baby's state detecting device, audio filter to filter baby's cry sound signal and to output interruption signal to the controller while detecting some frequency signal; video camera and transmitting unit. The present invention makes it possible for the nurse to know the baby's state while watching TV or operating computer.

The invention provides an acoustic system which is provided with an audible audio application module, an audible sound band audio signal mixer, an ultrasonic band processing module, an ultrasonic signal transmitter group, an adder, a digital / analog converter, a power amplification module, a loudspeaker, a microphone, a microphone signal amplification module, an analog / digital converter and an audible audio filter module. According to the acoustic system disclosed by the invention, the loudspeaker can be used for playing audio files and can also be used as a transmitting unit of the ultrasonic band processing module.

The present technology relates to a local sound field forming apparatus, a local sound field forming method, and a program that are capable of reducing sound leakage in an unintended direction.The local sound field forming apparatus includes a local sound field forming filter coefficient recording unit configured to record an audio filter coefficient for forming a sound field by an evanescent wave, a filter unit configured to convolve the audio filter coefficient and a sound source signal to generate a loudspeaker drive signal, and a loudspeaker array that includes a plurality of loudspeakers including a directional loudspeaker, and is configured to reproduce a sound on the basis of the loudspeaker drive signal. The present technology can be applied to a local sound field forming apparatus.

The invention discloses a real-time positioning recording device and a real-time positioning recording method; the real-time positioning recording device comprises a pickup matrix, a camera, a camera focus positioning and audio processing module, a multimedia file module, and a network sending module; the camera focus positioning and audio processing module comprises a camera positioning analysis module, a pickup audio filter module and an audio processing module; the pickup matrix comprises multiple pickups; after the camera starts to work, the camera positioning analysis module determines a focus, fallen on the pickup matrix, of the camera; the pickup matrix sends audio information to the pickup audio filter module and the audio processing module according to the focus; and the audio processing module respectively sends the processed audio file to the multimedia file module and the network sending module. According to the device and the method disclosed by the invention, a set of distributed pickup array is designed, thereby positioning a real-time pickup synthesizer according to the focus of the camera, determining sound synthesis collected by the pickup array in real time according to a position, fallen in a space, of the focus of the camera, and correlating to a video signal in real time.

A diode connected P-type CMOS transistor is operated in the sub-threshold area and, with a bypass capacitor, operates as a low pass audio filter. The equivalent resistance of the CMOS transistor in the sub-threshold range is very high—in the gigaOhm range. With this size resistor, a capacitor in the 1-25 pF range may be used to provide filtering capabilities with break points in the 1-10 Hz frequency range. Such a filter provides an effective low pass filter that attenuates audio frequency signals. The 1-25 pF capacitors use little chip area making the arrangement practical for integrating on an IC with the audio signals. In one embodiment, a digital signal and the audio signals share one pin, where the audio signal appears only when the digital signal is high. In this case, the audio signal filtered out from the digital high signal. The filtered digital signal drives digital circuitry while the audio signal is directed to other audio circuitry.

Provided is a method for outputting voice data in a portable terminal includes the steps of setting voice frequency bands of one language selected from a plurality of languages; modulating frequency bands of received voice data to the voice frequency bands of the set language; and outputting the voice data whose frequency bands are modulated to the voice frequency bands of the set language. Exemplary embodiments of the present invention allow for satisfactory conversation quality to be achieved on a portable terminal, regardless of the language of a talking partner, by setting an audio filter suitable for a language used by a user of the portable terminal.

The invention discloses an automobile tail door voice control circuit. The automobile tail door voice control circuit comprises a voice type-in module, an audio filtering module, a voice processing and recognizing module, an MCU control module, a tail door state detection circuit module and a tail door opening and closing control module, wherein the voice type-in module, the audio filtering module, the voice processing and recognizing module and the MCU control module are electrically connected in sequence; and the MCU control module is electrically connected with the tail door state detectioncircuit module and the tail door opening and closing control module. The invention further discloses a control method for the automobile tail door voice control circuit. The control method comprisesthe following steps: typing in voice signals; carrying out filtering processing and digital-to-analogue conversion so that audio data is generated; carrying out recognition and analysis on the audio data; and controlling the opening and closing of a tail door according to a voice instruction needing to be processed and the opening and closing information of the tail door. The electric tail door ofthe automobile is directly opened or closed through the voice command, and thus the reliability and safety for voice control of the automobile tail door are improved.

A media system and a method of using the media system to reproduce spatial audio based on head-to-torso orientation, are described. The method includes determining a head-to-source orientation and a head-to-torso orientation based on head orientation data generated by a head tracking device. Determining the head-to-torso orientation includes determining torso movements based on movements of the head. The torso can be determined to move when the head movements meet a head movement condition, such as a predetermined angle of movement or pattern of movement. A binaural audio filter that is based on a head-related transfer function corresponding to both the head-to-source orientation and the head-to-torso orientation is applied to an audio input signal to generate an audio output signal. The audio output signal is played to accurately recreate spatial audio having sounds emitted to the user by a sound source. Other aspects are also described and claimed.

The invention belongs to the field of the conference system, and discloses a videophone conference self-service guarantee control system. The system comprises a conference control system, a conferenceterminal system, a fault detection system, a control voice input module, a control voice input module, a control voice identification module and a cloud computation module; the conference control system comprises a control module, a conference establishment module, a member management module, a video information collection module, a video synthesis module, a video filtering module, a video sending module, an audio information collection module, an audio synthesis module, an audio filtering module, and an audio sending module; the conference control system comprises a video output module, an audio output module, a video display module and an audio playing module. The system disclosed by the invention is simple in structure, convenient for application, and convenient for maintenance, the conference efficiency is effectively improved, the conference cost is reduced, and the system has extensive application prospect.

A method of providing audio information, said method comprising: (i) receiving audio filter settings in a client device; (ii) receiving audio data in the client device, wherein the received audio data is in an audio coding format; (iii) converting the audio filter settings to an audio filter signal in a processor of the client device, where the audio filter signal is a time-varying signal; (iv) converting the received audio data to an audio signal in a processor of the client device, where the audio signal is a time-varying signal; and (v) transmitting the converted audio filter signal and the converted audio signal from the client device to an audio output device, where the audio output device is separate from and in communication with the client device, and where the audio output device is configured for modifying the audio signal according to the audio filter signal to generate a time-varying audio output.

An audio filter system including an audio notch filter having selected resistance and capacitance values, the system including up to ten parallel connected notch filters for a filter set, each notch filter selected to notch the same frequency or band of frequencies. A number of filter sets may further be connected to each other to notch or reject numerous different frequencies or band of frequencies as required by the application. The system filtering out frequencies and frequency bands not necessary for faithful reproduction of the original audio signal so as to substantially eliminate various forms of distortion, sideband loading and heterodyning.

The invention provides an audio filtering method, an audio filtering device and a wearable device. The audio filtering method is used for processing an original audio file recorded by the wearable device. The audio filtering method comprises the following steps: S101, carrying out fast Fourier transform on the original audio file, so that spectral characteristic data is obtained; S102, removing spectral characteristics corresponding to infrasound frequency and ultrasonic frequency from the spectral characteristic data; S103, determining an effective audio signal band, removing the spectral characteristics of the frequency band, except the effective audio signal band, in the spectral characteristic data, thus obtaining the spectral characteristic data with noise removed; and S104, carryingout Fourier inverse transform on the spectral characteristic data with noise removed, thus obtaining an audio file after filtering. After the technical scheme is implemented, the filtering processingis carried out on the original recording file on the wearable device, then the noise is removed, and thus the tone quality effect of the audio is promoted.

Embodiment apparatus and associated methods relate to automatically improving the quality of a segmented audio stream containing a desired signal, based on filtering an audio segment with an audio filter configured to remove a predetermined noise, and adapting the audio filter degree of noise removal determined as a function of a characteristic of the desired signal in the audio segment. In an illustrative example, the desired signal may be human voice. The predetermined noise may be, for example, wireless ring, hum, or tick resulting from a wireless microphone. In various implementations, the desired signal characteristic may be voice activity detected in the audio segment, and the audio filter degree of noise removal may be adapted as a function of the voice activity. Various examples may advantageously provide faster and more accurate vocal dialog editing in sound production procedures such as ADR (known as Alternative Dialog Replacement, or, Additional Dialog Recording).

The invention discloses a pre-amplification type novel recording system based on an audio power amplification circuit. The recording system is characterized by mainly comprising a microprocessor, a memory, a conversion module, an audio filter circuit, a timer, a pre-amplification circuit, an audio acquisition module, the audio power amplification circuit and a loudspeaker, wherein the memory, the conversion module, the audio filter circuit and the timer are respectively connected with the microprocessor; the pre-amplification circuit is connected with the conversion module; the audio acquisition module is connected with the pre-amplification circuit; the audio power amplification circuit is connected with the audio filter circuit; the loudspeaker is connected with the audio power amplification circuit; and the audio acquisition module is also connected with the microprocessor. The recording system has the advantages and beneficial effects that jamming noises can be inhibited and needed audio signals can be retained, so that the recording playing clarity can be improved; and the audio power amplification circuit is arranged and can be used for amplifying the power of the output audio signals, so that people can clearly listen to the recorded sounds repeatedly.

The invention discloses an analog-digital conversion type novel recording system based on a pre-amplification circuit. The analog-digital conversion type novel recording system is characterized by being mainly composed of a microprocessor, a storage, an analog-digital conversion circuit, an audio filtering circuit, a timer, the pre-amplification circuit, an audio collection module and a loudspeaker, wherein the storage, the analog-digital conversion circuit, the audio filtering circuit and the timer are connected with the microprocessor, the pre-amplification circuit is connected with the analog-digital conversion circuit, the audio collection module is connected with the pre-amplification circuit and the loudspeaker is connected with the audio filtering circuit. According to the analog-digital conversion type novel recording system, interference noise can be restrained, needed audio signals are reserved, and therefore the record playing definition can be improved. Due to the arrangement of the pre-amplification circuit, the pre-amplification circuit can conduct undistorted amplification on the collected audio signals, post-converted modulus and the microprocessor can conduct conversion and recognition on the audio signals, and therefore the stability of the analog-digital conversion type novel recording system is improved.