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 software defined radio detects an amplitude modulated (AM) signal contained within a received signal. This detection involves first receiving a radio frequency (RF) signal that contains the AM signal modulated about a carrier frequency (fc). The RF signal is downconverted using a first local oscillator having a frequency flo. An in-phase (I) channel signal and a quadrature (Q) channel signal are produced. From the I-channel signal, a relationship between the carrier frequency and the frequency of the local oscillator is determined with a frequency detector. The downconverted carrier signal of the I-channel signal is used to create a second set of signals with proper phases and frequencies that can be used to rotate the I-channel and Q-channel signals to account for differences between the downconverted fc and DC.

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 conference bridge. The method includes maintaining full intelligibility for all users even when there are multiple simultaneous talkers. The bridge also allows a user to hear a plurality of talkers over a single channel.

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: 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 an LPC coding standard to a bit stream representing frames of data according to a MELP coding standard. The method includes decoding a bit stream into a first set of parameters compatible with the LPC coding standard. Next, the LPC set of parameters are transformed into a MELP set of parameters compatible with the MELP coding standard without converting the LPC set of parameters to an analog or digital waveform representation. Lastly, the MELP set of parameters are encoded into a bit stream compatible with the MELP coding standard.

Abstract: The present invention comprises a system and method for correlation domain formant enhancement. The method includes attenuating an output speech signal in areas of low spectral amplitude. The attenuated signals are subsequently analyzed, and then converted to a format suitable for transmission.

Abstract: The system and method of the present invention comprises a compressed domain conference bridge. The method includes maintaining full intelligibility for all users even when there are multiple simultaneous talkers. The bridge also allows a user to hear a plurality of talkers over a single channel.

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 (MELP coding standard) to a bit stream representing frames of data according to a second compression standard (TDVC 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: The system and method of the present invention comprises a hybrid dual/single talker speech synthesizer that produces a digital audio stream. The method includes decoding a compressed bit stream that contains encoded parameters. The encoded parameters describe either a single talker's voice input or encoded parameters describing two simultaneous talker's voice inputs. The dual synthesizer is able to decode such dual talker bit streams and handle transitions from dual talker to single talker modes and vice versa without introducing objectionable artifacts in the output audio.

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 (MELP coding standard) to a bit stream representing frames of data according to a second compression standard (LPC 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.