Abstract: A CELP encoder is provided that optimizes excitation vector-related parameters in a more efficient manner than the encoders of the prior art. In one embodiment, a CELP encoder optimizes excitation vector-related parameters based on a computed correlation matrix, which matrix is in turn based on a filtered first excitation vector. The encoder then evaluates error minimization criteria based on at least in part on a target signal, which target signal is based on an input signal, and the correlation matrix and generates a excitation vector-related index in response to the error minimization criteria. In another embodiment, a CELP encoder is provided that is capable of jointly optimizing and/or sequentially optimizing multiple excitation vector-related parameters by reference to a joint search weighting factor, thereby invoking an optimal error minimization process.

Abstract: A speech coder that performs analysis-by-synthesis coding of a signal determines gain parameters for each constituent component of multiple constituent components of a synthetic excitation signal. The speech coder generates a target vector based on an input signal. The speech coder further generates multiple constituent components associated with the synthetic excitation signal, wherein one constituent component of the multiple constituent components is based on a shifted version of another constituent component of the multiple constituent components. The speech coder further evaluates an error criteria based on the target vector and the multiple constituent components to determine a gain associated with each constituent component of the multiple constituent components.

Abstract: To address the need for choosing values of harmonic noise weighting (HNW) coefficient (?p) so that the amount of harmonic noise weighting can be optimized, a method and apparatus for performing harmonic noise weighting in digital speech coders is provided herein. During operation, received speech is analyzed to determine a pitch period. HNW coefficients are then chosen based on the pitch period, and a perceptual noise weighting filter (C(z)) is determined based on the harmonic-noise weighting (HNW) coefficients (?p).

Abstract: A speech coder that performs analysis-by-synthesis coding of a signal determines gain parameters for each constituent component of multiple constituent components of a synthetic excitation signal. The speech coder generates a target vector based on an input signal. The speech coder further generates multiple constituent components associated with the synthetic excitation signal, wherein one constituent component of the multiple constituent components is based on a shifted version of another constituent component of the multiple constituent components. The speech coder further evaluates an error criteria based on the target vector and the multiple constituent components to determine a gain associated with each constituent component of the multiple constituent components.

Abstract: A CELP encoder is provided that optimizes excitation vector-related parameters in a more efficient manner than the encoders of the prior art. In one embodiment, a CELP encoder optimizes excitation vector-related parameters based on a computed correlation matrix, which matrix is in turn based on a filtered first excitation vector. The encoder then evaluates error minimization criteria based on at least in part on a target signal, which target signal is based on an input signal, and the correlation matrix and generates a excitation vector-related index in response to the error minimization criteria. In another embodiment, a CELP encoder is provided that is capable of jointly optimizing and/or sequentially optimizing multiple excitation vector-related parameters by reference to a joint search weighting factor, thereby invoking an optimal error minimization process.

Abstract: An open-loop delay contour estimator (204) generates delay information during coding of an information signal. The delay contour is adjusted according to an error minimization criterion on a subframe basis, which allows a more precise estimate of the true delay contour. A delay contour reconstruction block (211) uses the delay information in a decoder in reconstructing the information signal.

Abstract: To accurately determine rate and voice activity in moderate-to-low signal-to-noise ratios (SNRs) to maximize voice quality, system capacity and/or battery life, parameters from a noise suppression system are used as inputs to the rate determination function. Using this method, more of the speech is extracted from the background noise and a lower number of false onsets during fluctuating noise conditions compared with conventional systems are detected. The method is beneficial for voice activity detection (VAD) as well as rate determination (RDA) and unlike other RDA/VAD implementations, is independent of the type of speech coder employed (IS-127, CDG-27, IS-96 and GSM).

Abstract: Forward link communication capacity of a communication system, which is providing from a base station to a number of mobile stations a number of forward communication links corresponding to the plurality of mobile stations, is controlled by receiving gain setting associated with at least one of the forward communication links, comparing the gain setting with a gain threshold, and adjusting a first encoding rate of at least one of the forward communication links to a second encoding rate based on the comparing. A gain offset attunes the gain threshold to update the gain threshold according to a latest condition of the communication system. The gain offset further is used to determine blocking of an incoming call to the communication system.

Abstract: A mobile station (106) in a code division multiple access (CDMA) communication system (100) implements scanning of pilot channels on a different frequency by reserving a periodic frame on the forward channel to allow the mobile station (106) to change frequencies and scan for other pilot channels. To preserve some aspects of voice quality, both the base station (102) and mobile station (106) voice encode speech at a maximum of half rate and transmit the information as secondary traffic prior to the frame where the mobile station (106) scans the alternate frequency. To maximize the trade-off between voice quality and frequency of the scan, the base station (102) indicates to the mobile station (106) the period between other frequency scans via messaging. To ensure compatibility, this method can be negotiated via known service configuration negotiation techniques.

Abstract: A noise suppression system implemented in communication system provides an improved update decision during instances of sudden increase in background noise level. The noise suppression system, inter alia, generates an update by continually monitoring the deviation of spectral energy and forcing an update based on a predetermined threshold criterion. The spectral energy deviation is determined by utilizing an element which has the past values of the power spectral components exponentially weighted. The exponential weighting is a function of the current input energy, which means the higher the input signal energy the longer the exponential window. Conversely, the lower the signal energy the shorter the exponential window. The noise suppression system also inhibits a forced update during periods of continuous, non-stationary input signals (such as "music-on-hold").

Abstract: A method and apparatus are provided for echo canceling which includes double-talk immunity. The echo canceller (15) according to the present invention includes a double-talk detector (17) which uses an adaptive double-talk detection threshold (214) and an accurate estimation of reference energy (106) to detect double-talk. The echo canceller further provides for correcting partial echo canceller adaptive filter vector divergence occurring prior to double-talk detection by maintaining and switching to an auxiliary echo canceller filter with known "good" filter coefficients (58, 60, 62).

Abstract: Signaling is inserted (102) into a communication channel of a communication system by inserting the signaling into the communication channel if the communication channel has sufficient available bandwidth (104). If the communication channel does not have sufficient available bandwidth, reducing the bandwidth of information being communicated on the communication channel (106) and inserting the signaling into the communication channel (108) with reduced information bandwidth.

Abstract: A method is described of decreasing convergence time in an adaptive echo canceller. The method includes the step of locating a primary echo within a filter vector based upon relative tap values within the filter vector. The filter vector is then narrowed, based upon the located taps. An estimated error is determined based, in part, upon the narrowed filter vector. An updated filter vector is produced based, in part, upon the estimated error.

Abstract: An announcement and tone generator (10) for use with a telephonic network stores binarily encoded tone samples in a nonvolatile memory (22), such as a PROM or ROM, so that tones are immune to loss of memory due to power loss and, stores the binarily encoded announcements in a volatile memory, such as a DRAM, which is protected against memory loss by means of a back-up battery circuit. A microprocessor (20) provides successive samples of preselected binarily encoded tones or announcements for each of a plurality of time division multiplex channels which are coupled to the telephone network (12) through a digital interface (18).