Abstract: A system and method for detection of rate determination algorithm errors in variable rate communications system receivers. The disclosed embodiments prevent rate determination algorithm errors from causing audible artifacts such as screeches or beeps. The disclosed system and method detects frames with incorrectly determined data rates and performs frame erasure processing and/or memory state clean up to prevent propagation of distortion across multiple frames. Frames with incorrectly determined data rates are detected by checking illegal rate transitions, reserved bits, validating unused filter type bit combinations and analyzing relationships between fixed code-book gains and linear prediction coefficient gains.

Abstract: Methods and apparatus for quickly selecting an optimal excitation waveform from a codebook are presented herein. To reduce the number of computations required to choose the optimal codebook vector, a subset of codevectors are selected based upon optimal pulse locations, wherein the subset of codevectors form a subcodebook. Rather than searching the entire codebook, only the entries of the subcodebook are searched.

Abstract: The mobile telephone is provided with the capability for automatically adjusting the gain of a microphone of the telephone based upon the detected noise level in which the cellular telephone is operated. As the noise level increases, the gain of the microphone is automatically decreased, thereby compensating for the natural tendency of telephone users to speak more loudly in noisy environments. Also, by decreasing the microphone gain, any clipping that might otherwise occur as a result of the user speaking more loudly is avoided and the signal-to-noise ratio is not thereby decreased. Furthermore, because the microphone gain decreases, the volume level of the voice of the user as it is output from the other party's telephone is not unduly loud. Hence, the other party need not manually decrease the speaker gain of his or her telephone.

Abstract: Methods and apparatus for quickly selecting an optimal excitation waveform from a codebook are presented herein. In encoding schemes that use forward and backward pitch enhancement, storage and processor load is reduced by approximating a two-dimensional autocorrelation matrix with a one-dimensional autocorrelation vector. The approximation is possible when a cross-correlation element is configured to determine the autocorrelation matrix of an impulse response and a pulse energy determination element is configured to determine the energy of a pulse code vector that incorporates secondary pulse positions.

Abstract: The disclosed embodiments provide a method and apparatus for interoperability between CTX and DTX communications systems during transmissions of silence or background noise [FIG. 2]. Continuous eighth rate encoded noise frames are translated to discontinuous SID frames for transmission to DTX systems (402-410). Discontinuous SID frames are translated to continuous eighth rate encoded noise frames for decoding by a CTX system (602-606).

Abstract: A method and apparatus for enhancing coding efficiency by reducing illegal or other undesirable packet generation while encoding a signal. The probability of generating illegal or other undesirable packets while encoding a signal is reduced by first analyzing a history of the frequency of codebook values selected while quantizing speech parameters. Codebook entries are then reordered so that the index/indices that create illegal or other undesirable packets contain the least frequently used entry/entries. Reordering multiple codebooks for various parameters further reduces the probability that an illegal or other undesirable packet will be created during signal encoding. The method and apparatus may be applied to reduce the probability of generating illegal null traffic channel data packets while encoding eighth rate speech.

Abstract: The mobile telephone is provided with the capability for automatically adjusting the gain of a microphone of the telephone based upon the detected noise level in which the cellular telephone is operated. As the noise level increases, the gain of the microphone is automatically decreased, thereby compensating for the natural tendency of telephone users to speak more loudly in noisy environments. Also, by decreasing the microphone gain, any clipping that might otherwise occur as a result of the user speaking more loudly is avoided and the signal-to-noise ratio is not thereby decreased. Furthermore, because the microphone gain decreases, the volume level of the voice of the user as it is output from the other party's telephone is not unduly loud. Hence, the other party need not manually decrease the speaker gain of his or her telephone.

Abstract: A system and method for preparing and sending e-mail communications using a wireless communications device are disclosed. In one embodiment, input data comprising audio, image, and/or video data is encoded and transmitted to a cellular network. An integrated e-mail processor connected to the cellular network processes the coded data into a composite e-mail message and forwards the message to a server. The server then forwards the composite e-mail message to the indicated recipient(s). In another embodiment, the wireless communications device processes the coded data into a composite e-mail message and forwards it to a server via a cellular network. In either embodiment, the server may be dedicated to the cellular network. The invention thus enables users of handheld wireless communications devices, users of other devices lacking typing keyboards, or users presently unable to use a typing keyboard to prepare and send e-mail messages.

Abstract: The disclosed embodiments provide a method and apparatus for interoperability between CTX and DTX communications systems during transmissions of silence or background noise. Continuous eighth rate encoded noise frames are translated to discontinuous SID frames for transmission to DTX systems. Discontinuous SID frames are translated to continuous eighth rate encoded noise frames for decoding by a CTX system.

Abstract: Generally stated a method and an accompanying apparatus provides for a voice recognition system (300) with programmable front end processing unit (400). The front end processing unit (400) requests and receives different configuration files at different times for processing voice data in the voice recognition system (300). The configuration files are communicated to the front end unit via a communication link (310) for configuring the front end processing unit (400). A microprocessor may provide the front end configuration files on the communication link at different times.

Abstract: A speech processing system modifies various aspects of input speech according to a user-selected one of various preprogrammed voice fonts. Initially, the speech converter receives a formants signal representing an input speech signal and a pitch signal representing the input signal's fundamental frequency. One or both of the following may also be received: a voicing signal comprising an indication of whether the input speech signal is voiced, unvoiced, or mixed, and/or a gain signal representing the input speech signal's energy. The speech converter also receives user selection of one of multiple preprogrammed voice fonts, each specifying a manner of modifying one or more of the received signals (i.e., formants, voicing, pitch, gain). The speech converter modifies at least one of the formants, voicing, pitch, and/or gain signals as specified by the selected voice font.

Abstract: A voice recognition rejection scheme for capturing an utterance includes the steps accepting the utterance, applying an N-best algorithm to the utterance, or rejecting the utterance. The utterance is accepted if a first predefined relationship exists between one or more closest comparison results for the utterance with respect to a stored word and one or more differences between the one or more closest comparison results and one or more other comparison results between the utterance and one or more other stored words. An N-best algorithm is applied to the utterance if a second predefined relationship exists between the one or more closest comparison results and the one or more differences between the one or more closest comparison results and the one or more other comparison results.

Abstract: Methods and apparatus for quickly selecting an optimal excitation waveform from a codebook are presented herein. To reduce the number of computations required to choose the optimal codebook vector, a subset of codevectors are selected based upon optimal pulse locations, wherein the subset of codevectors form a subcodebook. Rather than searching the entire codebook, only the entries of the subcodebook are searched.

Abstract: A method and apparatus are described for controlling the data rates for communications to and from a base station and a plurality of remote users. The usage of the communications resource whether the forward link resource (from base station to remote users) or reverse link resource (from remote users to base station) is measured. The measured usage value is compared against at least one predetermined threshold value and the data rates of communications or a subset of communications on said communications resource is modified in accordance with said comparisons.

Abstract: A spoken user interface for speech-enabled devices includes a processor and a set of software instructions that are executable by the processor and stored in nonvolatile memory. A user of the speech-enabled device is prompted to enter a voice tag associated with an entry in a call history of the speech-enabled device. The call history includes lists of incoming and outgoing email messages, and incoming and outgoing telephone calls. The user is prompted to enter a voice tag after associated with a telephone number or email address in the call history after a user-selected number of telephone calls has been sent from the speech-enabled device to that telephone number, or has been sent from the telephone with that telephone number to the speech-enabled device, or after a user-selected number of email messages has been sent from the speech-enabled device to that email address, or has been sent from that email address to the speech-enabled device.

Abstract: Methods and apparatus for quickly selecting an optimal excitation waveform from a codebook are presented herein. In encoding schemes that use forward and backward pitch enhancement, storage and processor load is reduced by approximating a two-dimensional autocorrelation matrix with a one-dimensional autocorrelation vector. The approximation is possible when a cross-correlation element is configured to determine the autocorrelation matrix of an impulse response and a pulse energy determination element is configured to determine the energy of a pulse code vector that incorporates secondary pulse positions.

Abstract: According to a disclosed embodiment, an incoming call from a first telephone number is received (310). A variable indicating the number of calls received from the first telephone number is incremented (312). The variable is compared to a threshold variable (316). Subsequently, the user is prompted to save the first telephone number in an address book (319) when the variable indicating the number of times an incoming call is received from the first telephone number is equal to, or greater than, the threshold variable. The first telephone number is then stored in the address book. Moreover, an apparatus can be assembled (200) to provide cellular telephone users with, for example, instructions and information via a speaker (228) and an LCD display (238). The apparatus is configured to provide instructions and information to the cellular telephone user at appropriate times, thereby increasing the cellular telephone's usefulness.

Abstract: Wideband speech signals must be converted to narrowband speech signals if the transmission medium or the destination terminal is constructed with narrowband constraints. A typical wideband-to-narrowband conversion method is the elimination of frequencies above 3400 Hz using a low pass filter and a down sampler. However, this method produces a muffled speech sound since the resulting narrowband signal has a flat frequency response. Methods and apparatus are presented herein to enhance the acoustic quality of a wideband-to-narrowband converted signal. A bandwidth switching filter is used to emphasize a mid-range frequency portion of the wideband signal so that the resulting narrowband signal has a non-flat frequency spectrum.

Abstract: A method and apparatus for balancing the forward link capacity of a wireless communication system with the reverse link capacity of the system is presented. Speech coders with selectable modes are implemented in both links so that a forward link speech coder will not be operating with a mode set that is identical to the mode set used by a reverse link speech coder. Since the reverse link has a higher user capacity than a forward link, the reverse link speech coder can operate with a higher average data rate. Hence, the mode set used by the reverse link speech coder can be implemented without low average data rate modes. Elimination of modes from the mode set reduces complexity of the speech coder.

Abstract: It is an objective of the present invention to provide an optimized method of selection of the encoding mode that provides rate efficient coding of the input speech. It is a second objective of the present invention to identify and provide a means for generating a set of parameters ideally suited for this operational mode selection. Third, it is an objective of the present invention to provide identification of two separate conditions that allow low rate coding with minimal sacrifice to quality. The two conditions are the coding of unvoiced speech and the coding of temporally masked speech. It is a fourth objective of the present invention to provide a method for dynamically adjusting the average output data rate of the speech coder with minimal impact on speech quality.