Abstract:
An Integrated Development Environment (IDE) (100) for creating a touchless Virtual User Interface (VUI) 120 is provided. The IDE can include a development window (152) for graphically presenting a visual layout of user interface (UI) components (161) that respond to touchless sensory events in a virtual layout of virtual components (261), and at least one descriptor (121) for modifying a touchless sensory attribute of a user component. The touchless sensory attribute describes how a user component responds to a touchless touchless sensory event on a virtual component.
Abstract:
A device (100) and a method (200) for operating a camera (130) based on touchless movements is provided. The device (100) includes a sensing unit (110) for detecting a touchless movement, and a controller (130) for handling one or more controls of the camera in accordance with the touchless movement. A virtual user interface is provided to allow a user to control a camera on a computer or a mobile device using touchless finger movements. Touchless controls are provided for zoom, pan, focus, aperture, balance, color, calibration, or tilt. A first touchless finger movement can select a control, and a second touchless finger movement can adjust the control.
Abstract:
The invention concerns a system (100) and method (400) for operation of a voice activity detector (230). The system can include a speaker (105), a first microphone (110) and a second microphone (120) in which the first microphone and the second microphone can capture acoustic output from the speaker. The system can also include an adaptive module (220) in which the first microphone and the second microphone can provide signals to the adaptive module, and the adaptive module can provide an input to the voice activity detector. The adaptive module can receive a first input (242) from the first microphone and a second input (243) from the second microphone and can attempt to determine (430) a transformation between the first and second inputs for setting a configuration of the voice activity detector.
Abstract:
A speech filter (108) enhances the loudness of a speech signal by expanding the formant regions of the speech signal beyond a natural bandwidth of the formant regions. The energy level of the speech signal is maintained so that the filtered speech signal contains the same energy as the pre-filtered signal. By expanding the formant regions of the speech signal on a critical band scale corresponding to human hearing, the listener of the speech signal perceives it to be louder even though the signal contains the same energy.
Abstract:
A system, wireless device (102) and method improve speaker intelligibility in a multi-party call by receiving a plurality of individual voice signals, determining a pitch contour for each individual voice signal, determining that the pitch contours for at least two of the individual voice signals are within a predetermined range relative to each other, and shifting the pitch of at least one voice signal a predetermined amount for the duration of the call. The pitch of the individual voice is shifted one to approximately five semitones. The method is performed at a central control station (110) prior to summation of the signals, or at an individual receiving unit (204) when three or more wireless devices (102) are communicating without the use of a central control station (110).
Abstract:
Systems (100 or 300) and methods (400 or 500) are provided for selecting a post-compression waveform from a post-compression waveform table (106) and supplying it to a synthesis engine (108). The post-compression waveform is based upon a set of post-compression coefficients determined by generating a frequency-domain representation of a periodic signal, the representation including at least one pre-compression frequency-domain sample (204), and performing a threshold-based compression of the pre-compression frequency-domain samples. Systems and methods also include indexing and storing (502) post-compression coefficients in a post-compression coefficient table (102), generating (506) a post-compression waveform based upon the set of post-compression coefficients, and placing (508) the post-compression waveform in the table prior to the selecting (510). The system and method also include performing (504) a read-ahead operation on a sound file before selecting the post-compression waveform, the read-ahead operation indicating the post-compression waveform to be selected and supplied to the synthesis engine.
Abstract:
A method, system and computer readable medium for increasing the audio perceptual loudness includes shifting at least one frequency of a first audio signal to create a second audio signal so as to increase the audio perceptual loudness. The power level of the second audio signal is not more than a power level of the first audio signal. The method also includes generating high-audio perceptual loudness tone alert sequences based on psychoacoustic and audiometric data. It further includes acquiring a listener's threshold audio profile; adding the listener's audio profile to the loudness sensitivity curve for producing the listener's tonal sensitivity curve; determining a required dB scaling for critical band tones from the listener's tonal sensitivity curve; normalizing the tonal sensitivity curve for creating a decibel curve; selecting a frequency range of the tones by using the tonal sensitivity curve; and spacing the sequence of tones along a critical band scale.
Abstract:
A method of scaling polyphony can include identifying music data to be played (415), wherein the music data indicates instruments to be used and each instrument has an assigned priority. A measure of polyphony needed to play the music data can be compared with polyphony of a sound generating device (425). If the measure of polyphony exceeds the polyphony of the sound generating device, the music data can be played without using one or more instruments indicated by the music data according to the assigned priorities (440, 460).
Abstract:
A method (200) of voice identification within a mobile communication device can include detecting a voice signal within the mobile communication device (225) and determining at least one voice feature of the voice signal (230). At least one of the voice features of the voice signal can be compared with voice profiles accessible by the mobile communication device (235). Each voice profile also can be associated with an identity. Accordingly, at least one of the voice features of the voice signal can be matched with one of the voice profiles (240) and the identity associated with the matched voice profile can be presented through the mobile communication device (255).
Abstract:
A spine alignment system is provided to assess load forces on the vertebra in conjunction with overall spinal alignment. The system includes a spine instrument having an electronic assembly and a sensorized head. The sensorized head can be inserted between vertebra and report vertebral conditions such as force, pressure, orientation and edge loading. A GUI is therewith provided to show where the spine instrument is positioned relative to vertebral bodies as the instrument is placed in the inter-vetebral space. The system can distract vertebrae to a first height and measure the load applied by the spine region. The GUI can indicate that the load is outside a predetermined range. The spine region can be distracted to a second height where the load is measured within the predetermined load range.