77 results about "Voice Treatment" patented technology

A system for presenting a recorded message on behalf of an agent receiving a call at an agent station in a call center, the agent station having a telephony interface for receiving calls and a personal computer with a video display unit (PC / VDU), has a telephony switching apparatus enhanced by a computer-telephony integration (CTI) processor, an Interactive Voice Response (IVR) system executing a voice treatment option (VTO) software, coupled to the telephony switching apparatus and the CTI processor; and a data repository having recorded messages stored on behalf of the agent and accessible to the CTI processor. The CTI processor routes incoming calls to the telephony interface for the agent, uses data associated with the calls for selecting appropriate recorded messages, and causes, by controlling the IVR, a retrieved recorded message to be played to a caller upon the agent picking up the routed call. The system is useful with both conventional telephone systems and data network telephony, such as over the Internet.

An auditory prosthesis (30) comprising a microphone (27) for receiving the sound and producing a microphone signal responding to the received sound, an output device for providing audio signals in a form receivable by a user of the prosthesis (30), a sound processing unit (33) operable to receive the microphone signal and carry out a processing operation on the microphone signal to produce an output signal in a form suitable to operate the output device, wherein the sound processing unit (33) is operable in a first mode in which the processing operation comprises at least one variable processing factor which is adjustable by a user to a setting which causes the output signal of the sound processing unit (33) to be adjusted according to the preference of the user for the characteristics of the current acoustic environment.

An exemplary system includes 1) an electro-acoustic stimulation (“EAS”) sound processor configured to be located external to a patient, 2) a cochlear implant communicatively coupled to the EAS sound processor and configured to be implanted within the patient, 3) an electrode array communicatively coupled to the cochlear implant and configured to be located within a cochlea of the patient, and 4) a receiver communicatively coupled to the EAS sound processor and configured to be in communication with an ear of the patient. The EAS sound processor directs at least one of the cochlear implant and the receiver to apply stimulation to the patient, records an evoked response that occurs in response to the stimulation, and performs a predetermined action in accordance with the evoked response. Corresponding systems and methods are also disclosed.

Coding of received audio signals and the resulting application of electrical stimuli applied to electrodes used in a cochlear implant system are disclosed together with a method of fitting this new coding strategy. One of the aims is to improve place specific stimulation representing pitch by applying near threshold electrical stimuli with limited and focused excitation fields. A range of stimulation rates and a minimal range of current levels above threshold are used for creation of a dynamic loudness percept for a cochlear implant recipient. Another aim is to disclose a coding scheme based on a model of physiological measures (i.e. refractoriness, adaptation, spread of activation field, spatiotemporal acoustical cochlear activation patterns and spontaneous activity) to estimate the proportions of available excitable auditory neurons close to the electrodes available for stimulation. The spectral bands formed from the pre-processing of incoming audio signals are weighted by these proportions of excitability to control place, timing, rate and current level of electrical stimuli applied to the electrodes available in the array.

The embodiment of the invention discloses a method of processing telepresence conference user signals, which comprises: getting the signing in information of telepresence conference place and recognizing the free subscriber area according to the signing in information of users; making sure the signals from the free subscriber area are invalid signals and the signals from the other subscriber areas are valid signals. The invention also provides device and system thereof. According to the invention, in telepresence applied occasions, attending situation of users in each subscriber area can be recognized according to the simple and convenient signing in device arranged on the terminal side. And according to the information, in the processes of video switching, screen composing and acoustic processing, the invalid video and audio code stream can be effectively and automatically wiped off so as to improve the performance, efficiency and usability of conference system.

A headphone system includes sound processor which calculates properties of the environment from signals from an internal microphone and an external microphone. The impulse response of the environment may be calculated from the signals received from the internal and external microphones as the user speaks.

An exemplary sound processor (104) included in an auditory prosthesis system includes 1) a clock facility (406) configured to detect an elapsing of a predetermined amount of time and 2) a fitting facility (408) communicatively coupled to the clock facility and configured to automatically perform, in response to the elapsing of the predetermined amount of time, one or more fitting operations with respect to the auditory prosthesis system. Corresponding systems and methods are also disclosed.

The invention relates to equipment for fitting a hearing aid to the specific needs of a hearing impaired individual. The equipment comprising a computer or similar device with soft-ware, where the device has display means for visual display of data, data entry means for entering hearing aid programming data into the device, data storing means, and data output means for outputting programming data to the hearing aid. According to the invention means are provided for selecting simultaneous settings relating to two or more different parameters relating to the processing of sound in the hearing aid to be programmed.

The sound processor and method uses a model of basilar membrane motion to select stimuli, based upon the predicted motion which the acoustic signal presented would produce in an acoustically excited normally hearing cochlea. The filter; used, in contrast to single channel per electrode approaches, cover multiple channels and overlap with each other. Consequently the stimuli presented produce a neural excitation pattern which approximates the spatio-temporal travelling wave observed on the basilar membrane in an acoustically excited normally hearing cochlea. Preferably, the predicted electrode stimuli are based upon the instantaneous predicted amplitude of the electrode location.

The sound processing device, set between a sound source and a loud speaker and used for processing the sound output from the sound source and outputting it to the loud speaker, comprises: a selector for selecting an singing- exercising mode or a normal mode; an original sound eliminating mode for attenuating the human sound frequency in the audio signals output by the sound source under singing-excising mode and a tone regulation mode for regulating the tone of the audio signals output from the sound source under singing-excising mode. Advantage: raised convenience and amusement for the user of excising singing.

The invention discloses a baby crying sound translation method based on sound feature recognition. According to crying sound characteristic differences of babies in different physiological states, crying sound characteristic parameters of babies in different physiological states are extracted respectively by using a computer sound processing technology, wherein the crying sound characteristic parameters are 16 characteristic parameters including 11 characteristic parameters: the tone, pitch, loudness, energy, frequency and frequency co-occurrence matrix; on the basis of a BP neural network algorithm, crying sound sections of the babies in different physiological states are collected, denoising and filtering are carried out on the sound sections, a correspondence relationship between baby crying sound characteristic differences and six kinds of physiological states including hunger, drowsiness, pain, boredom, fear and discomfort is established, and identification results of hunger, drowsiness, pain, boredom, fear and discomfort are provided. In the trained BP neural network, characteristic parameters extracted from any baby crying sound section are inputted and thus an identification result is obtained from the output layer, so that the baby care quality and efficiency are improved and the erroneous determination and delayed determination in baby caring are reduced.