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: A voice recognition (VR) system is disclosed that utilizes a combination of speaker independent (SI) and speaker dependent (SD) acoustic models. At least one SI acoustic model is used in combination with at least one SD acoustic model to provide a level of speech recognition performance that at least equals that of a purely SI acoustic model. The disclosed hybrid SI/SD VR system continually uses unsupervised training to update the acoustic templates in the one or more SD acoustic models.

Abstract: A call setup procedure is presented to permit vocoder bypass, which will allow the transmission of wideband speech packets between wideband terminals over narrowband transmission constraints. In addition, methods and apparatus are presented that allow the conversion between a wideband tandem-free operation, a narrowband tandem-free operation, and a standard tandem operation.

Abstract: A method and apparatus providing a user interface within a phone that responds to a limited vocabulary of user trained voice commands. The interface allows users to perform all phone handset dialing functions using voice commands. Additionally, users will be able to create and modify entries within a voice recognition phonebook, whereby a number within the voice recognition phonebook can be called by saying the name associated with the number. The user interface provides a combination of voice and LCD displayed user prompts and responses to voice input. The interface responds to user voice commands and performs the command functions based upon matches to previously user trained voice command vocabulary words stored in memory.

Abstract: A method and apparatus for using coding scheme selection patterns in a predictive speech coder to reduce sensitivity to frame error conditions includes a speech coder configured to select from among various predictive coding modes. After a predefined number of speech frames have been predictively coded, the speech coder codes one frame with a nonpredictive coding mode or a mildly predictive coding mode. The predefined number of frames can be determined in advance from the subjective standpoint of a listener. The predefined number of frames may be varied periodically. An average coding bit rate may be maintained for the speech coder by ensuring that an average coding bit rate is maintained for each successive pattern, or group, of predictively coded speech frames including at least one nonpredictively coded or mildly predictively coded speech frame.

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: A method and apparatus for identifying frequency bands to compute linear phase shifts between frame prototypes in a speech coder includes partitioning the frequency spectrum of a prototype of a frame by dividing the frequency spectrum into segments, assigning one or more bands to each segment, and establishing, for each segment, a set of bandwidths for the bands. The bandwidths may be fixed and uniformly distributed in any given segment. The bandwidths may be fixed and non-uniformly distributed in any segment. The bandwidths may be variable and non-uniformly distributed in any given segment.

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: A method and system for voice recognition combines different types of voice recognition (VR) engines in order to recognize user-defined digits and control words, predefined digits and control words, and nametags. Speaker-independent VR engines are combined with speaker-dependent VR engines. A Hidden Markov Model (HMM) VR engine is combined with Dynamic Time Warping (DTW) engines.

Abstract: A novel and improved method and an accompanying apparatus provide for a distributed voice recognition (VR) capability in a remote device (201). Remote device (201) decides and controls what portions of the VR processing may take place at remote device (201) and what other portions may take place at a base station (202) in wireless communication with remote device (201).

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: An apparatus for testing user interface integrity of speech-enabled devices includes a processor and a storage medium coupled to the processor. A set of voiced utterances is stored in the storage medium. A software module is executed by the processor to determine a state of the voice recognizer and provide a response to the voice recognizer in accordance with the determined state. The response may be to produce at least one voiced utterance in accordance with the state. The apparatus may be acoustically coupled to the voice recognizer. The apparatus may also, or in the alternative, be electrically coupled by a cable to the voice recognizer. The set of voiced utterances may include multiple sets of voiced utterances, each set having been spoken by a different person. The set of voiced utterances may also, or in the alternative, include multiple sets of voiced utterances, each set of voiced utterances having been spoken under different background noise conditions.

Abstract: A method and apparatus for maintaining a target bit rate in a speech coder includes a speech coder for encoding a frame at a preselected encoding rate, computing a running average bit rate for a predefined number of encoded frames, subtracting the running average bit rate from a predefined target average bit rate, and dividing the difference by the preselected encoding rate. If the quotient value is negative, a predefined number of possible occurrence counts of speech coder performance threshold values that are less than a current performance threshold value is accumulated, the accumulated number being greater than the absolute value of the quotient. The product of a decrement-per-occurrence-count-value and the predefined number of occurrence counts is subtracted from the current performance threshold value to obtain a new performance threshold value.

Abstract: An apparatus for accurate endpointing of speech in the presence of noise includes a processor and a software module. The processor executes the instructions of the software module to compare an utterance with a first signal-to-noise-ratio (SNR) threshold value to determine a first starting point and a first ending point of the utterance. The processor then compares with a second SNR threshold value a part of the utterance that predates the first starting point to determine a second starting point of the utterance. The processor also then compares with the second SNR threshold value a part of the utterance that postdates the first ending point to determine a second ending point of the utterance. The first and second SNR threshold values are recalculated periodically to reflect changing SNR conditions. The first SNR threshold value advantageously exceeds the second SNR threshold value.

Abstract: A method and apparatus for providing feedback from the decoder to the encoder to improve performance in a predictive speech coder under frame erasure conditions includes notifying an encoder in a receiving speech coder if a decoder in the receiving speech coder fails to receive a frame transmitted by an encoder in a transmitting speech coder. A modified packet is transmitted from the encoder in the receiving speech coder to a decoder in the transmitting speech coder in response to the notification. The decoder in the transmitting speech coder notifies the encoder in the transmitting speech coder when the modified packet is received. The encoder in the transmitting speech coder then encodes a packet with a modified encoding format. The modified encoding format may be either a low-memory encoding format or a memoryless encoding format. The modified packet may have an erasure indicator bit set to a digital value of one.

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.

Abstract: A method and apparatus for CELP-based to CELP-based vocoder packet translation. The apparatus includes a formant parameter translator and an excitation parameter translator. The formant parameter translator includes a model order converter and a time base converter. The method includes the steps of translating the formant filter coefficients of the input packet from the input CELP format to the output CELP format and translating the pitch and codebook parameters of the input speech packet from the input CELP format to the output CELP format. The step of translating the formant filter coefficients includes the steps of converting the model order of the formant filter coefficients from the model order of the input CELP format to the model order of the output CELP format and converting the time base of the resulting coefficients from the input CELP format time base to the output CELP format time base.

Abstract: A method and apparatus for CELP-based to CELP-based vocoder packet translation. The apparatus includes a formant parameter translator and an excitation parameter translator. The formant parameter translator includes a model order converter and a time base converter. The method includes the steps of translating the formant filter coefficients of the input packet from the input CELP format to the output CELP format and translating the pitch and codebook parameters of the input speech packet from the input CELP format to the output CELP format. The step of translating the formant filter coefficients includes the steps of converting the model order of the formant filter coefficients from the model order of the input CELP format to the model order of the output CELP format and converting the time base of the resulting coefficients from the input CELP format time base to the output CELP format time base.

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.

Abstract: The mobile telephone detects bad data packets by decoding speech parameters represented by the packets and comparing those speech parameters against acceptable ranges of speech parameters. If the speech parameters do not lie within the acceptable ranges, the packets are discarded. This speech parameter-based detection of bad data packets is particularly useful in mobile telephones receiving variable rate data packets wherein an error in determining the rate of a packet results in the entire packet being decoded at an incorrect rate, thereby likely causing an annoying audible artifact in the decode speech signal. Such rate detection errors can occur even though a CRC or other conventional error detection check fails to detect the bad packet. In an exemplary implementation, the mobile telephone receives signals encoded with TIA/EIA/IS-95-A standards.