Abstract: At an audio source, pause information is added to audio data, the combination of which is subsequently packetized. The resulting packets are transmitted to an audio destination via a network in which different packets may be subjected to varying levels of delay. At the audio destination, the pause information may be used to insert pauses at appropriate times to accommodate the occurrence of delays in packet delivery. In one embodiment, pauses are inserted based on a hierarchy of pause types. During pauses, audio filler information may be injected. In this manner, the effects of variable network delays upon reconstructed audio may be mitigated.

Abstract: In a two-way radio communications system wherein subscriber units are provided with multiple features from the software running on a server, a user interface of the subscriber unit permits the subscriber unit user to change the control state of the server software depending on different types of actuations of the user interface. In the preferred embodiment, one, two or more “clicks” on a momentary input switch, as well as pushing and holding and releasing the input switch can be used to select or de-select between multiple services and applications provided to the subscriber unit by the server software.

Abstract: In a wireless communication system, local detection of an interrupt indicator during a voice communication between a user of a subscriber unit and another person is provided. Responsive to the interrupt indicator, a portion of a speech recognition element is activated to begin processing voice-based commands. The speech recognition element can be implemented at least in part within an infrastructure, such as in a client-server speech recognition arrangement. The interrupt indicator may be provided using an input device forming a part of the subscriber unit, or through the use of a local speech recognizer within the subscriber unit. By locally detecting interrupt indicators at the subscriber unit, the present invention more readily enables the use of electronic assistants and similar services in wireless communication environments.

Abstract: At an audio source, pause information is added to audio data, the combination of which is subsequently packetized. The resulting packets are transmitted to an audio destination via a network in which different packets may be subjected to varying levels of delay. At the audio destination, the pause information may be used to insert pauses at appropriate times to accommodate the occurrence of delays in packet delivery. In one embodiment, pauses are inserted based on a hierarchy of pause types. During pauses, audio filler information may be injected. In this manner, the effects of variable network delays upon reconstructed audio may be mitigated.

Abstract: A start of an input speech signal is detected during presentation of an output audio signal and an input start time, relative to the output audio signal, is determined. The input start time is then provided for use in responding to the input speech signal. In another embodiment, the output audio signal has a corresponding identification. When the input speech signal is detected during presentation of the output audio signal, the identification of the output audio signal is provided for use in responding to the input speech signal. Information signals comprising data and/or control signals are provided in response to at least the contextual information provided, i.e., the input start time and/or the identification of the output audio signal. In this manner, the present invention accurately establishes a context of an input speech signal relative to an output audio signal regardless of the delay characteristics of the underlying communication system.

Abstract: At an audio source, pause information is added to audio data, the combination of which is subsequently packetized. The resulting packets are transmitted to an audio destination via a network in which different packets may be subjected to varying levels of delay. At the audio destination, the pause information may be used to insert pauses at appropriate times to accommodate the occurrence of delays in packet delivery. In one embodiment, pauses are inserted based on a hierarchy of pause types. During pauses, audio filler information may be injected. In this manner, the effects of variable network delays upon reconstructed audio may be mitigated.

Abstract: A wireless system comprises at least one subscriber unit in wireless communication with an infrastructure. Each subscriber unit implements a speech recognition client, and the infrastructure comprises a speech recognition server. A given subscriber unit takes as input an unencoded speech signal that is subsequently parameterized by the speech recognition client. The parameterized speech is then provided to the speech recognition server that, in turn, performs speech recognition analysis on the parameterized speech. Information signals, based in part upon any recognized utterances identified by the speech recognition analysis, are subsequently provided to the subscriber unit. The information signals may be used to control the subscriber unit itself; to control one or more devices coupled to the subscriber unit, or may be operated upon by the subscriber unit or devices coupled thereto.

Abstract: A start of an input speech signal is detected during presentation of an output audio signal and an input start time, relative to the output audio signal, is determined. The input start time is then provided for use in responding to the input speech signal. In another embodiment, the output audio signal has a corresponding identification. When the input speech signal is detected during presentation of the output audio signal, the identification of the output audio signal is provided for use in responding to the input speech signal. Information signals comprising data and/or control signals are provided in response to at least the contextual information provided, i.e., the input start time and/or the identification of the output audio signal. In this manner, the present invention accurately establishes a context of an input speech signal relative to an output audio signal regardless of the delay characteristics of the underlying communication system.

Abstract: In a two-way radio communications system wherein subscriber units are provided with multiple features from the software running on a server, a user interface of the subscriber unit permits the subscriber unit user to change the control state of the server software depending on different types of actuations of the user interface. In the preferred embodiment, one, two or more “clicks” on a momentary input switch, as well as pushing and holding and releasing the input switch can be used to select or de-select between multiple services and applications provided to the subscriber unit by the server software.

Abstract: At an audio source, pause information is added to audio data, the combination of which is subsequently packetized. The resulting packets are transmitted to an audio destination via a network in which different packets may be subjected to varying levels of delay. At the audio destination, the pause information may be used to insert pauses at appropriate times to accommodate the occurrence of delays in packet delivery. In one embodiment, pauses are inserted based on a hierarchy of pause types. During pauses, audio filler information may be injected. In this manner, the effects of variable network delays upon reconstructed audio may be mitigated.

Abstract: An input speech signal is encoded as one or more reflection coefficients. To reduce storage requirements, the reflection coefficients are scalar quantized by storing an N-bit code rather than the entire reflection coefficient. An exemplary value for N is 8. A table is provided having 2.sup.N reflection coefficient values. The N-bit code is used to look up reflection coefficient values from the table. To reduce spectral distortion due to scalar quantization, the reflection coefficient values in the table are non-linearly scaled.

Abstract: An improved speech coder provides a more natural sounding replication of speech by modifying the mean-squared error criterion for the selected speech coder parameters. Specifically, the modification emphasizes the signal components that the speech coder has difficulty matching, i.e. the high frequencies. This emphasis is constrained to certain limitations to avoid over-emphasizing the speech.

Abstract: A Vector-Sum Excited Linear Predictive Coding (VSELP) speech coder provides improved quality and reduced complexity over a typical speech coder. VSELP uses a codebook which has a predefined structure such that the computations required for the codebook search process can be significantly reduced. This VSELP speech coder uses single or multi-segment vector quantizer of the reflection coefficients based on a Fixed-Point-Lattice-Technique (FLAT). Additionally, this speech coder uses a pre-quantizer to reduce the vector codebook search complexity and a high-resolution scalar quantizer to reduce the amount of memory needed to store the reflection coefficient vector codebooks. Resulting in a high quality speech coder with reduced computations and storage requirements.

Abstract: A digital speech coding method uses an Rth-order filter to model the frequency response of multiple filters, thereby, providing a filter which offers the control of multiple filters without the complexity of multiple filters. The Rth-order filter can be used as a spectral noise weighting filter or a combination of a short-term predictor filter and a spectral noise weighting filter, referred to as the spectrally noise weighted synthesis filter, depending on which embodiment is employed. In general, the method models the frequency response of L Pth-order filters by a single Rth-order filter, where the order R<L.times.P. Thus, this method increases the control of a speech coder filter without a corresponding increase in the complexity of the speech coder.

Abstract: A digital speech coder utilizes harmonic noise weighting to overcome some limitations of low-rate CELP-type speech coders in reproducing voiced speech. In addition to a short term correction factor, which constitutes spectral noise weighting as known in the art, a long term pitch correction factor is utilized to provide harmonic noise weighting. The inclusion of harmonic noise weighting in a speech coder more efficiently utilizes noise-masking properties of a speech signal, allowing synthesis of a higher quality speech at a given bit rate.

Abstract: A speech coder and decoder methodology wherein pitch excitation and codebook excitation source energies are represented by parameters that are readily transmissible with minimal transmission capacity requirements. The parameters are the long term energy value, a short term correction factor which is applied to the long term energy value to match the short term energy, and proportionality factor(s) that specify the relative energy contribution of the excitation sources to the short term energy value.

Abstract: An Rth-order filter models the frequency response of multiple filters, to provide a filter which offers the control of multiple filters without the complexity of multiple filters. The Rth-order filter can be used as a spectral noise weighting filter or a combination of a short-term predictor filter and a spectral noise weighting filter, referred to as the spectrally noise weighted synthesis filter, depending on which embodiment is employed. In general, the method models the frequency response of L Pth-order filters by a single Rth-order filter, where the order R<L.times.P. Thus, this method increases the control of a speech coder filter without a corresponding increase in the complexity of the speech coder.

Abstract: A digital speech coder includes a long-term filter (124) having an improved sub-sample resolution long-term predictor (FIG. 5 ) which allows for subsample resolution for the lag parameter L. A frame of N samples of input speech vector s(n) is applied to an adder (510). The output of the adder (510) produces the output vector b(n) for the long term filter (124). The output vector b(n) is fed back to a delayed vector generator block (530) of the long-term predictor. The nominal long-term predictor lag parameter L is also input to the delayed vector generator block (530). The long-term predictor lag parameter L can take on non-integer values, which may be multiples of one half, one third, one fourth or any other rational fraction. The delayed vector generator (530) includes a memory which holds past samples of b(n).

Abstract: A wireless comprises at least one subscriber unit in wireless communication with an infrastructure. Each subscriber unit implements a speech recognition client, and the infrastructure comprises a speech recognition server. A given subscriber unit takes as input an unencoded speech signal that is subsequently parameterized by the speech recognition client. The parameterized speech is then provided to the speech recognition server that, in turn, performs speech recognition analysis on the parameterized speech. Information signals, based in part upon any recognized utterances identified by the speech recognition analysis, are subsequently provided to the subscriber unit. The information signals may be used to control the subscriber unit itself; to control one or more devices coupled to the subscriber unit, or may be operated upon by the subscriber unit or devices coupled thereto.

Abstract: A wireless system comprises at least one subscriber unit in wireless communication with an infrastructure. Each subscriber unit implements a speech recognition client, and the infrastructure comprises a speech recognition server. A given subscriber unit takes as input an unencoded speech signal that is subsequently parameterized by the speech recognition client. The parameterized speech is then provided to the speech recognition server that, in turn, performs speech recognition analysis on the parameterized speech. Information signals, based in part upon any recognized utterances identified by the speech recognition analysis, are subsequently provided to the subscriber unit. The information signals may be used to control the subscriber unit itself; to control one or more devices coupled to the subscriber unit, or may be operated upon by the subscriber unit or devices coupled thereto.