174 results about "Sound analysis" patented technology

Apparatus for use detection of apnea includes a microphone mounted in the ear of the patient for detecting breathing sounds and a second external microphone together with an oximetric sensor. A transmitter at the patient compresses and transmits the signals to a remote location where there is provided a detector module for receiving and analyzing the signals to extract data relating to the breathing. The detector uses the entropy or range of the signal to generate an estimate of air flow while extracting extraneous snoring and heart sounds and to analyze the estimate of air flow using Otsu's threshold to detect periods of apnea and/or hypopnea. A display provides data of the detected apnea/hypopnea episodes and related information for a clinician.

A lung sound diagnostic system for use in collecting, organizing and analyzing lung sounds associated with the inspiration(s) and expiration(s) of a patient. The system includes a plurality of transducers that may be placed at various sites around the patient's chest. The microphones are coupled to signal processing circuitry and A/D converters which digitize the data and preferably provides the digital data to a computer station. A data collection and organization program, executing on the computer station, organizes and formats the data into a combination display for display or printing. The combinational display includes at least two display elements. In a first display element, the data is shown for both inspiration and expiration combined in a first time scale. In a second display element, the data for inspiration and expiration are shown individually in a second time scale that is time-expanded relative to the first time scale. The system may also include application programs for detecting and classifying abnormal sounds. The resulting information may be displayed in a variety of formats to facilitate diagnosis. Additionally, the system may include an analysis program for comparing selected criteria corresponding to the detected abnormal sounds with predefined thresholds in order to provide a likely diagnosis.

The present invention relates to a method and system for analysing audio (eg. music) tracks. A predictive model of the neuro-physiological functioning and response to sounds by one or more of the human lower cortical, limbic and subcortical regions in the brain is described. Sounds are analysed so that appropriate sounds can be selected and played to a listener in order to stimulate and/or manipulate neuro-physiological arousal in that listener. The method and system are particularly applicable to applications harnessing a biofeedback resource.

Apparatus for use detection of apnea includes a microphone mounted in the ear of the patient for detecting breathing sounds and a second external microphone together with an oximetric sensor. A transmitter at the patient compresses and transmits the signals to a remote location where there is provided a detector module for receiving and analyzing the signals to extract data relating to the breathing. The detector uses the entropy or range of the signal to generate an estimate of air flow while extracting extraneous snoring and heart sounds and to analyze the estimate of air flow using Otsu's threshold to detect periods of apnea and/or hypopnea. A display provides data of the detected apnea/hypopnea episodes and related information for a clinician.

A sound analysis apparatus stores sound source structure data defining a constraint on one or more of sounds that can be simultaneously generated by a sound source of an input audio signal. A form estimation part selects fundamental frequencies of one or more of sounds likely to be contained in the input audio signal with peaked weights from various fundamental frequencies during sequential updating and optimizing of weights of tone models corresponding to the various fundamental frequencies, so that the sounds of the selected fundamental frequencies satisfy the sound source structure data, and creates form data specifying the selected fundamental frequencies. A previous distribution imparting part imparts a previous distribution to the weights of the tone models corresponding to the various fundamental frequencies so as to emphasize weights corresponding to the fundamental frequencies specified by the form data created by the form estimation part.

Methods of using individually distinctive patterns of voice characteristics to identify a speaker include computing the reassigned spectrogram of each of at least two voice samples, pruning each reassigned spectrogram to remove noise and other computational artifacts, and comparing (either visually or with the aid of a processor) the strongest points to determine whether the voice samples belong to the same speaker.

An information processing apparatus is provided which includes a beat analysis unit for detecting positions of beats included in an audio signal, a structure analysis unit for calculating similarity probabilities, each being a probability of similarity between contents of sound of beat sections divided by each beat position detected by the beat analysis unit, and a chord progression detection unit for determining a likely chord progression of the audio signal based on chord probabilities determined according to the similarity probabilities calculated by the structure analysis unit, each chord probability being a probability of each type of chord for each beat section.

A 2.3 kb/s speech coding technique with harmonic excitation and linear prediction is disclosed. The speech signals are acquired by speech acquisition system and then inputted to the processor of computer. The fundational tone detecting method and voiced sound analysis method are used to split the LPC exciting signal into lower band associated with voised sound and higher band relative to voiceless sound. A united harmonic model with a fixed phase spectrum of voiced sound of man speech is used to express the harmonic component and pseudo-noise component. An interpolation method is used to reduce noise. A short-lag filter is used for high quality of synthetic speech.

The invention relates to a medical monitoring system (100) based on sound analysis in a medical environment. A sound level analyzer (SLA, 10) is capable of providing an indicator for perceived levels of sound from a number of sound events, and a data storage modality (DSM, 20) is receiving and storing said indicator for perceived levels of sound and also corresponding information from an associated patient monitoring system (PMS, 60) handling information indicative of a physical and/or mental condition of a patient under influence by sound. A sound event analyzer (SEA, 30) is further being arranged for performing, within a defined time window, an overall sound analysis (ANA, 50) related to physical and/or mental condition of the patient that may be influenced by sound in order to assist or supervise medical personal with respect to the acoustic environment.

A method includes receiving, from a portable recording device, multiple recordings of sounds made by an elevator during operation of the elevator, the multiple recordings generated at different times. The method includes analyzing, by a sound analysis circuit, the multiple recordings generated at the different times to detect changes in an elevator characteristic over time and generating, based on detecting the changes in the elevator characteristic over time, a notification to perform an action on the elevator.

The invention provides a human-computer interaction song singing system and a human-computer interaction song singing method. The human-computer interaction song singing system comprises a server module, a play module, a display module, a sound acquisition module, a sound output module and a sound analysis module; the server module is used for storing audio and video files and subtitle files of songs; the play module is used for acquiring the audio and video files and the subtitle files from the server module; the sound acquisition module is used for acquiring singing audio information of a singer; the sound analysis module is used for analyzing the singing audio information into singing tones and outputting the singing tones obtained by analysis to the display module; the display module is used for receiving and displaying the video files, the subtitle files, the audio files and the singing tones; and the sound output module is used for playing an audio of a song and an audio of the singer. By adopting the method, the problems that in the prior art, the human-computer interaction effect is poor and the scoring cannot be carried out by intonation can be solved.

The invention provides an artificial intelligence video interview method and system. According to the intelligent video interview method and system, a video interview mode is adopted; expression analysis, sound analysis and analysis on answers of interview questions in the interview process are provided for job seekers through an artificial intelligence technology; the expression ability, the expression method, the matching degree with the post and the like of the job seeker can be automatically, accurately, comprehensively and efficiently evaluated; according to the method, the employer can be helped to screen the applicant according to the comprehensive evaluation result, the job seeker can fully understand the advantages and disadvantages in the interview process through feedback of comprehensive evaluation, pertinence is improved, the job seeker is helped to improve the quality of the job seeker, and the interview success rate is increased.

A voice analysis device includes: an acoustic parameter calculation unit which collects a voice signal and calculates an acoustic parameter for each time unit; a category decision unit which decides one of environment sound categories to which a partial voice signal of the unit time belongs; a section setting unit which sets from time to time, a judgment object section including a plurality of continuous time units along the time axis as the time elapses; and an environment judgment unit which judges the environment surrounding a local device in the judgment object section according to the number of environment sound categories contained in the judgment object section at least in the vicinity.

Disclosed are a method for determining whether a person is drunk after consuming alcohol on the basis of a difference among a plurality of formant energy energies, which are generated by applying linear predictive coding according to a plurality of linear prediction orders, and a recording medium and a terminal for carrying out the method. The alcohol consumption determining terminal comprises: a voice input unit for receiving voice signals and converting same into voice frames and outputting the voice frames; a voiced/unvoiced sound analysis unit for extracting voice frames corresponding to a voiced sound from among the voice frames; an LPC processing unit for calculating a plurality of formant energy energies by applying linear predictive cording according to the plurality of linear prediction orders to the voice frames corresponding to the voiced sound; and an alcohol consumption determining unit for determining whether a person is drunk after consuming alcohol on the basis of a difference among the plurality of formant energy energies which have been calculated by the LPC processing unit, thereby determining whether a person is drunk after consuming alcohol depending on a change in the formant energy energies generated by applying linear predictive coding according to the plurality of linear prediction orders to voice signals.

A converter transformer operating state on-line audio analyzing and monitoring system belongs to the field of monitoring devices. The system comprises a sound collection unit, a sound transmission unit, a sound analysis unit, a historical sound library, and an analysis result comparison and analysis unit, which are sequentially connected. The sound collecting unit collects actual operating sounds of a converter transformer and converts the sounds into electrical signals. The sound transmission unit transmits the electrical signals converted from the collected actual operating sounds of the converter transformer to the sound analysis unit. The sound analysis unit conducts audio decomposition of the electrical signals converted from the collected actual operating sounds of the converter transformer and forms typical amplitude and frequency curves of the converter transformer on different operating conditions. The historical sound library stores the typical amplitude and frequency curves of the converter transformer on different operating conditions for comparison and comparative analysis. The analysis result comparison and analysis unit is used for analysis and comparison of newly collected audios of the converter transformer, analysis and comparison of historical sounds, and automatic alarming when the collected operating sounds of the converter transformer is abnormal.

The invention discloses a sound based volume adjustment device. The device comprises a signal processing module, a noise acquisition module, a gain module, a pattern base, an input module and an output module. The noise acquisition module is used for acquiring a noise signal, the signal processing module performs sound analytical calculation according to an input audio signal and the noise signal, and the gain module is used for automatically adjusting volumes. The invention further provides a sound based audition product, and the sound based volume adjustment device is arranged in the product. The invention further provides a sound based volume adjusting method, and the method includes the steps of step one, playing input audio and acquiring the noise signal; step two, performing sound analysis if the noise signal exceeds a preset valve value; and step three, automatically adjusting the volume of the played audio.

Golf performance and equipment characteristics may be determined by analyzing the impact between a golf ball and an impacting surface. In some examples, the impacting surface may be a golf club face. The impact between the golf ball and the surface may be measured based on sound and/or motion sensors (e.g., gyroscopes, accelerometers, etc.). Based on motion and/or sound data, various equipment-related information including golf ball compression, club head speed and impact location may be derived. Such information and/or other types of data may be conveyed to a user to help improve performance, aid in selecting golf equipment and/or to insure quality of golfing products.

Golf performance and equipment characteristics may be determined by analyzing the impact between a golf ball and an impacting surface. In some examples, the impacting surface may be a golf club face. The impact between the golf ball and the surface may be measured based on sound and/or motion sensors (e.g., gyroscopes, accelerometers, etc.). Based on motion and/or sound data, various equipment-related information including golf ball compression, club head speed and impact location may be derived. Such information and/or other types of data may be conveyed to a user to help improve performance, aid in selecting golf equipment and/or to insure quality of golfing products.