Abstract: A method and apparatus that manages speech decoders in a communication device may include detecting a change in transmission rate from a higher rate to a lower rate, decoding and shifting a first, second and third received first decoder set of frame parameters, generating a first decoder output audio frame from the previously shifted frame parameters, generating a first, second and third second decoder audio fill frame, the second decoder being a higher rate decoder than first decoder, outputting a first and second second decoder audio fill frame, combining the first decoder audio frame and the third second decoder audio fill frame with overlapping triangular windows, and outputting combined first decoder and second decoder frames to an audio buffer for subsequent transmission to a user of the communication device. In an alternative embodiment, another method may include detecting and processing a change in transmission rate from a lower rate to a higher rate.

Abstract: A method and apparatus that generates fill frames for Voice over Internet Protocol (VoIP) applications in a communication device is disclosed.

Abstract: A method and apparatus that controls a Voice over Internet Protocol (VoIP) decoder in a communication device is disclosed. The method may include determining if a packet has been received, and if a packet has been received, determining if a receive error has occurred, and if a receive error has not occurred, setting a counter that counts a number of sequential bad frames to a value of zero, decoding the received packet, and sending the decoded packet to an audio queue for presentation to a user of the communication device.

Abstract: A method for transcoding a bit stream encoded according to a mixed-excitation linear prediction (MELP) standard to a bit stream encoded according to a linear predictive coding (LPC) standard, including: decoding a bit stream into a first set of vocoder parameters compatible with the MELP standard; transforming the first set of vocoder parameters into a second set of vocoder parameters compatible with the LPC standard without converting the first set of vocoder parameters to an analog or digital waveform representation; and encoding the second set of vocoder parameters into a bit stream compatible with the LPC vocoder standard.

Abstract: A method and apparatus that manages speech decoders in a communication device is disclosed. The method may include detecting a change in transmission rate from a higher rate to a lower rate, decoding and shifting a first, second and third received first decoder set of frame parameters, generating a first decoder output audio frame from the previously shifted frame parameters, generating a first, second and third second decoder audio fill frame, the second decoder being a higher rate decoder than first decoder, outputting a first and second second decoder audio fill frame, combining the first decoder audio frame and the third second decoder audio fill frame with overlapping triangular windows, and outputting combined first decoder and second decoder frames to an audio buffer for subsequent transmission to a user of the communication device. In an alternative embodiment, another method may include detecting and processing a change in transmission rate from a lower rate to a higher rate.

Abstract: A method and apparatus that generates fill frames for Voice over Internet Protocol (VoIP) applications in a communication device is disclosed.

Abstract: A method and apparatus that controls a Voice over Internet Protocol (VoIP) decoder in a communication device is disclosed. The method may include determining if a packet has been received, and if a packet has been received, determining if a receive error has occurred, and if a receive error has not occurred, setting a counter that counts a number of sequential bad frames to a value of zero, decoding the received packet, and sending the decoded packet to an audio queue for presentation to a user of the communication device.

Abstract: A method for transcoding a bit stream encoded according to a linear predictive coding (LPC) standard to a bit stream encoded according to a mixed-excitation linear prediction (MELP) standard, including: decoding a bit stream into a first set of vocoder parameters compatible with the LPC standard; transforming the first set of vocoder parameters into a second set of vocoder parameters compatible with the MELP standard without converting the first set of vocoder parameters to an analog or digital waveform representation; and encoding the second set of vocoder parameters into a bit stream compatible with the MELP vocoder standard.

Abstract: A method for enhancing speech prediction coefficient signals, including: a) receiving a set of prediction coefficient signals; b) calculating a set of amplitude spectrum coefficient signals based on the prediction coefficient signals; c) calculating a set of enhancement coefficient signals based on the amplitude spectrum coefficient signals; d) calculating a set of enhanced autocorrelation coefficient signals based on the enhancement coefficient signals; e) converting at least some of the enhanced autocorrelation coefficient signals into reflection coefficient signals; f) testing the range of values of the reflection coefficient signals; and g) converting at least some of the tested reflection coefficient signals into prediction coefficient signals.

Abstract: The system and method of the present invention comprises a compressed domain voice activity detector that detects the presence or absence of voice activity in a digital input signal. The method includes converting a digital input signal into parametric data. The parametric data is subsequently analyzed, and then compared against a background noise threshold to determine if voice activity is present.

Abstract: The system and method of the present invention comprises a compressed domain universal transcoder that transcodes a bit stream representing frames of data encoded according to a first compression standard (TDVC coding standard) to a bit stream representing frames of data according to a second compression standard (MELP coding standard). The method includes decoding a bit stream into a first set of parameters compatible with a first compression standard. Next, the first set of parameters are transformed into a second set of parameters compatible with a second compression standard without converting the first set of parameters to an analog or digital waveform representation. Lastly, the second set of parameters are encoded into a bit stream compatible with the second compression standard.

Abstract: An integrated full duplex radio data link for aircraft capable of a sustained 4 to 8 Mb/sec raw data transfer rate communicates on newly proposed dedicated digital channels (DDC) for radio communications between the existing analog VHF amplitude modulated (AM) voice channels, and causes no perceptible interference to the remaining AM voice channels. While the present invention is intended primarily for terminal areas and airport surface applications, it may be employed with enroute applications. The present invention transmits AM encoded digital aircraft identification information appended to AM voice transmissions. An ground unit strips off the aircraft identification and assigns a DDC frequency to that aircraft unit over a control uplink channel (CUC) for further data communication. Downlink information, such as acknowledgement of the DDC assigned is transmitted over a digital control downlink channel (CDC). Data communication then proceeds on the DDC.

Abstract: Hybrid subband coding and decoding (using different encloding/decoding algorithms in at least one subband channel) and subband time delay compensation at a point of maximum digital bandwidth compression are effected so as to reduce required DSP on-chip memory requirements. At the same time a special digital signal format is employed so as to provide enhanced data frame synchronization, enhanced cryptographic synchronization and selective signalling ability within a cryptographic digital signal transceiver. The ability to successfully accomplish late entry (or to re-establish synchronization once lost) into an ongoing received message is provided.

Abstract: A method for pitch-aligned high frequency regeneration of a speech signal which has been sampled at a known sampling frequency f.sub.S and decimated at a known integer decimation ratio N practiced in the receiver portion of a RELP vocoder includes the steps of: providing at least one local carrier signal(s), (each) at a frequency which is an exact integer multiple of a baseband pitch estimate frequency recovered from received data; amplitude modulating each of the local carrier signals with baseband residual data recovered in the receiver portion to provide partial spectrum data; removing, only if the decimation ratio is even, the lower sideband data from the lowest frequency local carrier signal to obtain partial spectrum data; and adding the residual baseband data to the partial spectrum data to obtain PA-HFRed output data from which to reconstruct the speech signal.

Abstract: A two-input crosstalk-resistant adaptive noise canceller receives a primary input signal including a desired speech signal portion and an undesired noise signal portion and also receives a reference input signal having a reference noise input portion and a crosstalk speech portion. The canceller has first and second summer means and first and second adaptive filter means. The first summer means provides a canceller output signal which is the difference between the primary input signal and the first adaptive filter output signal. The canceller output signal is applied to the reference input of the second adaptive filter and to one of a pair of error-control inputs of the first adaptive filter. The second error-control input of the first adaptive filter is provided by the signal at the output of the second adaptive filter, which receives a single error-control input signal from the output of the second summer means.