Abstract: Determination of a set of acoustic parameters for a headset is presented herein. The set of acoustic parameters can be determined based on a virtual model of physical locations stored at a mapping server. The virtual model describes a plurality of spaces and acoustic properties of those spaces, wherein the location in the virtual model corresponds to a physical location of the headset. A location in the virtual model for the headset is determined based on information describing at least a portion of the local area received from the headset. The set of acoustic parameters associated with the physical location of the headset is determined based in part on the determined location in the virtual model and any acoustic parameters associated with the determined location. The headset presents audio content using the set of acoustic parameters received from the mapping server.

Abstract: Systems, devices, and methods are described for moving a patient to and from various locations, care units, etc., within a care facility. For example a transport and support vehicle includes a body structure including a plurality of rotatable members operable to frictionally interface the vehicle to a travel path and to move the vehicle along the travel path, and a surface structured and dimensioned to support an individual subject. A transport and support vehicle can include, for example, an imager operably coupled to one or more of a power source, a steering assembly, one or more of the plurality of rotatable members, etc., and having one or more modules operable to control the power source, steering assembly, one or more of the plurality of rotatable members, etc., so as to maintain an authorized operator in the image zone.

Abstract: The invention relates to audio signal processing. More specifically, the invention relates to enhancing multichannel audio, such as television audio, by applying a gain to the audio that has been smoothed between portions of the audio. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods.

Abstract: A method of pre-processing an audio signal transmitted to a user terminal via a communication network and an apparatus using the method are provided. The method of pre-processing the audio signal may prevent deterioration of a sound quality of the audio signal transmitted to the user terminal by pre-processing the audio signal, and by enabling a codec module, encoding the audio signal, to determine the audio signal as a speech signal. Also, the method of pre-processing the audio signal may improve a probability that the codec module may determine a corresponding audio signal as a speech when the audio signal is transmitted via the communication network by pre-processing the audio signal using a speech codec.

Abstract: A noise estimation apparatus includes a correlation calculator configured to calculate a correlation value of a spectrum between a plurality of frames in sound information obtained using one or more microphones, a power calculator configured to calculate a power value indicating a sound level of one target frame among the plurality of frames, an update determiner configured to determine an update degree indicating a degree to which the sound information of the target frame is to be reflected in a noise model stored in a storage, or determine whether or not the noise model is to be updated to another noise model, based on the power value of the target frame and the correlation value, and an updater configured to generate the other noise model based on a determined result, the sound information of the target frame, and the noise model.

Abstract: In order to improve the speech quality of an electric larynx (EL) speaker, the speech signal of which is digitized by suitable means, the following steps are carried out: a) dividing a single-channel speech signal into a series of frequency channels by transferring it from a time domain into a discrete frequence domain; b) filtering out the modulation frequency of the EL by way of a high-pass or notch filter, in each frequency channel; and c) back-transforming the filtered speech signal from the frequency domain into the time domain and combining it into a single-channel output signal.

Abstract: An apparatus and a method for improving communication sound quality in a mobile terminal in order to remove a neighboring noise that occurs together with a user's voice signal in a mobile terminal by discriminating signals occurring at different distances using two microphones and removing a noise. The mobile terminal preferably includes a first microphone, a second microphone, and a voice processor. The first microphone receives a voice signal occurring at a closer distance from the mobile terminal and a voice signal occurring at a longer distance from the mobile terminal. The second microphone receives only a voice signal occurring at the long distance. The voice processor discriminates between the signal occurring at the long distance and the signal occurring at the close distance by receiving voice signals received via the first microphone and the second microphone at different phases.

Abstract: A communication device includes memory, an input interface, a processing module, and a transmitter. The processing module receives a digital signal from the input interface, wherein the digital signal includes a desired digital signal component and an undesired digital signal component. The processing module identifies one of a plurality of codebooks based on the undesired digital signal component. The processing module then identifies a codebook entry from the one of the plurality of codebooks based on the desired digital signal component to produce a selected codebook entry. The processing module then generates a coded signal based on the selected codebook entry, wherein the coded signal includes a substantially unattenuated representation of the desired digital signal component and an attenuated representation of the undesired digital signal component. The transmitter converts the coded signal into an outbound signal in accordance with a signaling protocol and transmits it.

Abstract: A portable golf GPS device comprising a microprocessor operably coupled to a GPS unit, an input device such as a keypad (or touch screen) operably coupled to the microprocessor, a voice recognition unit operably coupled to the microprocessor, and a display such as a liquid crystal display (“LCD”) operably coupled to the microprocessor. A program memory system which contains at least some of the software and data to operate the device is also operably coupled to the microprocessor. The portable golf GPS device is preferably contained in a housing such that the entire device has a very compact and lightweight form factor, and is preferably handheld and/or pocket size. The golf GPS device is configured to display distances to course features, and to receive and process voice input in order to select functions or input data into the golf GPS device.

Abstract: A method and apparatus for continuously improving the performance of semantic classifiers in the scope of spoken dialog systems are disclosed. Rule-based or statistical classifiers are replaced with better performing rule-based or statistical classifiers and/or certain parameters of existing classifiers are modified. The replacement classifiers or new parameters are trained and tested on a collection of transcriptions and annotations of utterances which are generated manually or in a partially automated fashion. Automated quality assurance leads to more accurate training and testing data, higher classification performance, and feedback into the design of the spoken dialog system by suggesting changes to improve system behavior.

Abstract: In a method of smoothing background noise in a telecommunication speech session; receiving and decoding S1O a signal representative of a speech session, the signal comprising both a speech component and a background noise component. Subsequently, determining LPC parameters S20 and an excitation signal S30 for the received signal. Thereafter, synthesizing and outputting (S40) an output signal based on the determined LPC parameters and excitation signal. In addition, modifying S35 the determined excitation signal by reducing power and spectral fluctuations of the excitation signal to provide a smoothed output signal.

Abstract: A speech aid for persons with hypokinetic dysarthria, a speech disorder associated with Parkinson's disease. The speech aid alters the pitch at which the user hears his or her voice and/or provides multitalker babble noise to the speaker's ears. The speech aid induces increased speech motor activity and improves the intelligibility of the user's speech. The speech aid may be used with a variety of microphones, headphones, in one or both ears, with a voice amplifier, or connected to telephones.

Abstract: In one alternative, an audio signal is analyzed using multiple psychoacoustic criteria to identify a region of the signal in which time scaling and/or pitch shifting processing would be inaudible or minimally audible, and the signal is time scaled and/or pitch shifted within that region. In another alternative, the signal is divided into auditory events, and the signal is time scaled and/or pitch shifted within an auditory event. In a further alternative, the signal is divided into auditory events, and the auditory events are analyzed using a psychoacoustic criterion to identify those auditory events in which the time scaling and/or pitch shifting processing of the signal would be inaudible or minimally audible. Further alternatives provide for multiple channels of audio.

Abstract: In order to limit the range of externally outputable content of externally input speech, an MFP includes: a speech acquiring portion to acquire externally input speech; a speech converting portion to convert the acquired speech into character information; a user extracting portion to extract user identification information for identifying a user from the character information; and an output control portion to output the character information based on the extracted user identification information.

Abstract: The invention describes a computer-based system that asks (101) a patient to pronounce a word displayed on a monitor, automatically assesses (104, 105) the speech quality, and uses suitable means to feed back (106) any improvement or deterioration of speech quality.

Abstract: A method and machine-readable medium for providing virtual spatial sound with an audio visual player are disclosed. Input audio is processed into output audio having spatial attributes associated with the spatial sound represented in a room display.

Abstract: A method for audio modulation is provided. The method including: obtaining the digital audio signals of the caller in the process of communications; analyzing the digital audio signals and obtain a voice frequency of the caller; reading a voice frequency of the user from a memory, and calculate the rate of the voice frequencies between the caller and the user; modulating the user' analog audio signals according to the rate of the voice frequencies; converting the modulated analog audio signals into digital audio signals; coding the digital audio signals and modulating the coded digital audio signals and transmitting the modulated digital audio signals to the caller. Through the method, the user's voice is modulated to sound like the caller's voice, thereby increasing the interest of the process of communicating. Present invention also provides a communication device with the function of audio modulation.

Abstract: Sound signals from sound sources present in multiple directions are accepted as inputs of multiple channels, and signal of each channel is transformed into a signal on a frequency axis. A phase component of the transformed signal is calculated for each identical frequency, and phase difference between the multiple channels is calculated. An amplitude component of the transformed signal is calculated, and a noise component is estimated from the calculated amplitude component. An SN ratio for each frequency is calculated on the basis of the amplitude component and the estimated noise component, and frequencies at which the SN ratios are larger than a predetermined value are extracted. Difference between arrival distances is calculated on the basis of the phase difference at selected frequency, and the arrival direction in which it is estimated that the target sound source is present is calculated.

Abstract: Sound quality and speech comprehensibility are to be improved for hearing device wearers when watching television. Provision is made for this purpose to record acoustic signals at a recording site on a data medium. The acoustic signals are recorded simultaneously with a first microphone as a first recording. The data medium is played back on an individual playback device in an individual environment. Here the signal played back from the data medium is re-recorded as a second recording with a second microphone. The two recordings are connected to each other, subtracted in particular, and the result is used to adjust a hearing device program. It is thus possible to take into account individual acoustic environmental conditions in the hearing device program.