Abstract: A new method for quantization of the LPC coefficients in a speech coder includes a new weighted error measure including every frame sampling an impulse response from LPC filter 21 of said coder, filtering the samples using a perceptual weighting filter 39 and processing in a computer 39 to calculate autocorrelation function of the weighted impulse response, computing Jacobian matrix for LSF (Line Spectral Frequency), computing correlation of rows of Jacobian matrix and calculating LSF weights by multiplying correlation matrices.

Abstract: An audio processing apparatus is constructed for generating an auxiliary audio signal based on an original audio signal and mixing the auxiliary audio signal to the original audio signal. In the apparatus, a control section designates a pitch of the auxiliary audio signal. A processing section processes the original audio signal under control of the control section to generate the auxiliary audio signal having the designated pitch, and applies a first effect to the generated auxiliary audio signal. An effector section applies a second effect different from the first effect to the original audio signal. An output section outputs the original audio signal applied with the second effect concurrently with the auxiliary audio signal applied with the first effect. The control section may control the processing section to alter the first effect dependently on a difference between a pitch of the original audio signal and the designated pitch of the auxiliary audio signal.

Abstract: A method populates fields of a form. A form has fields and predefined standard responses associated with at least a subset of the fields. Each predefined standard response is associated with a distinct field. A text stream is received, a particular command in the text stream is identified. In response to the particular command, at least one field is populated with the predefined standard response associated with the field.

Abstract: A speech enhancement method, including the steps of: (a) segmenting an input speech signal into a plurality of frames and transforming each frame signal into a signal of the frequency domain; (b) computing the signal-to-noise ratio of a current frame, and computing signal-to-noise ratio of a frame immediately preceding the current frame; (c) computing the predicted signal-to-noise ratio of the current frame which is predicted based on the preceding frame and computing the speech absence probability using the signal-to-noise ratio and predicted signal-to-noise ratio of the current frame; (d) correcting the two signal-to-noise ratios obtained in the step (b) based on the speech absence probability computed in the step (c); (e) computing the gain of the current frame with the two corrected signal-to-noise ratios obtained in the step (d), and multiplying the speech spectrum of the current frame by the computed gain; (f) estimating the noise and speech power for the next frame to calculate the predicted signal-to-

Abstract: A system transmits two audio services, each related to a different video program, within the bandwidth typically reserved for a single audio channel. Each audio service may be processed as a mono, stereo or encrypted signal prior to modulation onto its own audio carrier. The two audio carriers are summed together and applied to an RF modulator. The frequencies of the audio carriers are selected to ensure that the quality of the video program is not degraded, the video channel mask bandwidth is not exceeded and interference from the unselected audio carrier is negligible.

Abstract: The present invention has the purpose of realizing an infallible in-call incoming signal notification incoming signal notification service with this erroneous action eliminated to the extent possible.