Abstract: Preprocessing speech signals from an indirect conduction microphone. One exemplary method preprocesses the speech signal in two stages. In stage one, an external speech sample is characterized using an auto regression model, and coefficients from the model are convolved with the internal speech signal from the indirect conduction microphone to produce a pre-conditioned internal speech signal. In stage two, a training sound is received by the indirect conduction microphone and filtered through a low-pass filter. The result is then modeled using auto regression, and inverted to produce an inverted filter model. The pre-conditioned internal speech signal is convolved with the inverted filter model to remove negative or undesirable acoustic characteristics and loss from the speech signal from the indirect conduction microphone.

Abstract: A method and apparatus for enhancing modulation of certain speech sounds, such as trill sounds, are provided for radios which utilize digital vocoders. A digitized speech stream is sampled and the sampling is adjusted to determine, detect and enhance trill nulls in the digitized voice stream by one or more of: frame shifting the digitized speech input stream prior to vocoding, time expanding a digitized speech steam prior to vocoding, time compressing a digitized speech output stream after vocoding, and/or modulation enhancement and filtering of the a digitized speech output stream after vocoding.

Abstract: Preprocessing speech signals from an indirect conduction microphone. One exemplary method preprocesses the speech signal in two stages. In stage one, an external speech sample is characterized using an auto regression model, and coefficients from the model are convolved with the internal speech signal from the indirect conduction microphone to produce a pre-conditioned internal speech signal. In stage two, a training sound is received by the indirect conduction microphone and filtered through a low-pass filter. The result is then modeled using auto regression, and inverted to produce an inverted filter model. The pre-conditioned internal speech signal is convolved with the inverted filter model to remove negative or undesirable acoustic characteristics and loss from the speech signal from the indirect conduction microphone.

Abstract: A method, a subscriber unit, and a system provide concurrent voice and data systems and methods that “steal bits” from voice frames for low-speed concurrent data. In this manner, concurrent voice and data is supported over protocols that require voice-only in current operation such as Project 25 (P25). The stealing of bits from voice is done in a manner that is transparent and not noticeable to users. The systems and methods enable replication of signaling on the downlink and wireline segments that is replaced on the uplink, encoding of interleaving blocks that extend across multiple Layer 2 voice logical data units (LDUs), flexibility in replacing voice data in LDUs, indication of the availability of the service and an interleaving schedule, notification of upcoming stealing events via signaling, and the like.

Abstract: A method and apparatus for enhancing modulation of certain speech sounds, such as trill sounds, are provided for radios which utilize digital vocoders. A digitized speech stream is sampled and the sampling is adjusted to determine, detect and enhance trill nulls in the digitized voice stream by one or more of: frame shifting the digitized speech input stream prior to vocoding, time expanding a digitized speech steam prior to vocoding, time compressing a digitized speech output stream after vocoding, and/or modulation enhancement and filtering of the a digitized speech output stream after vocoding.

Abstract: A method, a subscriber unit, and a system provide concurrent voice and data systems and methods that “steal bits” from voice frames for low-speed concurrent data. In this manner, concurrent voice and data is supported over protocols that require voice-only in current operation such as Project 25 (P25). The stealing of bits from voice is done in a manner that is transparent and not noticeable to users. The systems and methods enable replication of signaling on the downlink and wireline segments that is replaced on the uplink, encoding of interleaving blocks that extend across multiple Layer 2 voice logical data units (LDUs), flexibility in replacing voice data in LDUs, indication of the availability of the service and an interleaving schedule, notification of upcoming stealing events via signaling, and the like.

Abstract: A policy enforcement device performs a method for adjusting video compression parameters for encoding source video based on a viewer's environment. The method includes: receiving, from a video receiving device, a video stream identifier indicating a video stream and a source of the video stream, wherein the video receiving device is remote from the source of the video stream; receiving an indication of environmental conditions of a viewer of the video stream using the video receiving device; determining a set of video compression parameters based on the indication of environmental conditions of the viewer; sending the set of video compression parameters to the identified source of the video stream to enable encoding of the video stream to a different compression level.

Abstract: A policy enforcement device performs a method for adjusting video compression parameters for encoding source video based on a viewer's environment. The method includes: receiving, from a video receiving device, a video stream identifier indicating a video stream and a source of the video stream, wherein the video receiving device is remote from the source of the video stream; receiving an indication of environmental conditions of a viewer of the video stream using the video receiving device; determining a set of video compression parameters based on the indication of environmental conditions of the viewer; sending the set of video compression parameters to the identified source of the video stream to enable encoding of the video stream to a different compression level.

Abstract: A system and method for effectively and reliably terminating a voice call in any burst within a multi-burst superframe. A transmitting unit generates a termination burst upon detecting a dekey event. The termination burst includes a data synchronization pattern, a slot type field indicating an end of a call, and an information field surrounding the data synchronization pattern and the slot type field. The information field is encoded from a predetermined voice encoder frame unique to the termination burst. Once all buffered voice information is transmitted, the termination burst is transmitted prior to the end of the multi-burst superframe. A base station or other receiving unit monitors the incoming signal. Upon detecting the data synchronization pattern, the receiving unit decodes the slot type field and the information field.

Abstract: A system and method for effectively and reliably terminating a voice call in any burst within a multi-burst superframe. A transmitting unit generates a termination burst upon detecting a dekey event. The termination burst includes a data synchronization pattern, a slot type field indicating an end of a call, and an information field surrounding the data synchronization pattern and the slot type field. The information field is encoded from a predetermined voice encoder frame unique to the termination burst. Once all buffered voice information is transmitted, the termination burst is transmitted prior to the end of the multi-burst superframe. A base station or other receiving unit monitors the incoming signal. Upon detecting the data synchronization pattern, the receiving unit decodes the slot type field and the information field.

Abstract: The following describes an apparatus for and method of providing and supporting distributed call handling over a plurality of network nodes (401, 403, 405, 407, 409, 411, 413) in a communication system, wherein the network nodes (401, 403, 405, 407, 409, 411, 413) are found in each device with the system, rather than only in the repeaters (101, 103, 105, 107, 109), the site controller (111), or the zone controller (ZC) for the system. The call establishment process is provided as elements (301, 303, 305, 307, 309, 311, 313) of a state machine. Each of the elements (301, 303, 305, 307, 309, 311, 313) is distributed over one or more of the plurality of network nodes (401, 403, 405, 407, 409, 411, 413). If one of the network nodes becomes unavailable, the element(s) supported by the now unavailable network node is reassigned (705, 809), as necessary.

Abstract: Within a communication system, updating encryption key information in communication units of a particular communication group begins when a key management terminal transmits a key update message for the particular communication group to a communication resource allocator. Upon receipt of the key update message, the communication resource allocator determines a rekey session number from the key update message and allocates a communication resource for the particular communication group. Each active communication unit of the particular communication group receives the rekey session number and compares it to a stored rekey session number. If the numbers do not substantially match, the communication unit affiliates with the allocated communication resource to receive updated encryption key information from the key management terminal.