Abstract: A system and method are provided for creating shorter more natural sounding voice messages and prompts from a plurality of pre-recorded sound segments, the prerecorded sound segments are dynamically cross faded in order to produce a more natural blended sound, various cross fade parameters such as the fade length and the shape of the cross fade amplitude envelopes are determined based on characteristics of the various sound segments being combined.

Abstract: A system allocates channel bandwidth based on the data received from a plurality of remote sources. A de-multiplexer/priority circuit separates two or more different data streams into their components parts. A stream modification driver modifies one or more characteristics of the data received from the de-multiplexer/priority circuit based on a priority assigned to the data by the de-multiplexer/priority circuit. The de-multiplexer/priority circuit determines the data transfer rates for each of the different data streams based on the assigned priority.

Abstract: A system synchronizes data flow between a first device and a second device. The system includes a data link that connects two or more devices that are capable of sending and receiving data through a bus. A capture device senses and transfer information through the bus. A ring buffer temporarily stores data transmitted through the bus. A read controller copies or reconstructs data in a length that is different from the length of the data received. A monitor detects underflow or overflow conditions into or out of the ring buffer and compensates for clock drift.

Abstract: A speech enhancement system that improves the intelligibility and the perceived quality of processed speech includes a frequency transformer and a spectral compressor. The frequency transformer converts speech signals from the time domain to the frequency domain. The spectral compressor compresses a pre-selected portion of the high frequency band and maps the compressed high frequency band to a lower band limited frequency range. The speech enhancement system may be built into, may be a unitary part of, or may be configured to interface other systems that process audio or high frequency signals.

Abstract: A voice reinforcement system extracts a portion of a converted speech signal and redirects it towards a listening area where it may be added with the original signal. The system includes a speech input, a filter, and a converter. The speech input generates an intermediate signal from a speech signal. The filter extracts a portion of the signal extending above a cutoff frequency. The converter converts the filtered signal to an aural signal directed towards a listening area.

Abstract: A system processes signals exchanged between a near end terminal and a far end terminal over a communication path. The system selects the processing characteristics of a near end terminal based on characteristics of the communication path. The communication path may include the near and the far end terminal and their respective codecs, and the networks. The system may select processing characteristics of the near end terminal based on characteristics of the communication path. Selecting the processing characteristics may include selecting a predetermined operation. The system may adapt a selected operation based on a real time analysis of the communication path.

Abstract: An automatic gain control system maintains desired signal content level, such as voice, in an output signal. The system includes automatic gain control over an input signal, and compensates the output signal based on input signal content. When the input signal level exceeds an upper or lower processing threshold level, or is distorted (e.g., clipped), the system applies a gain to the input signal level. The system may compensate for the gain in the output signal when the input signal includes desired signal content.

Abstract: A rule-based end-pointer isolates spoken utterances contained within an audio stream from background noise and non-speech transients. The rule-based end-pointer includes a plurality of rules to determine the beginning and/or end of a spoken utterance based on various speech characteristics. The rules may analyze an audio stream or a portion of an audio stream based upon an event, a combination of events, the duration of an event, or a duration relative to an event. The rules may be manually or dynamically customized depending upon factors that may include characteristics of the audio stream itself, an expected response contained within the audio stream, or environmental conditions.

Abstract: A system and method are provided for creating shorter more natural sounding voice messages and prompts from a plurality of pre-recorded sound segments, the prerecorded sound segments are dynamically cross faded in order to produce a more natural blended sound, various cross fade parameters such as the fade length and the shape of the cross fade amplitude envelopes are determined based on characteristics of the various sound segments being combined.

Abstract: A processing system generates an output signal which includes desired signal components, and reduces or eliminates tonal noise. The output signal may be provided to any subsequent signal processing system, including voice recognition systems, pitch detectors, and other processing systems. The subsequent processing systems are less likely to mistake tonal input signal noise for desired signal content, to needlessly consume computational resources to analyze noise, and to take spurious actions induced by the tonal noise.

Abstract: The operation of a vehicle electronic module is influenced by vehicle-state received on a vehicle communication bus. The electronic module analyzes vehicle-state data that includes information about the vehicle's environment. The electronic module improves or maintains electronic module performance by adjusting communication processing parameters to improve driver or passenger communication.