Abstract: The present disclosure is directed towards a method for scheduling data packets in a multi-channel packet processing environment. The method may include receiving one or more data packets associated with an incoming signal and inserting the one or more data packets into a queue. The method may further include monitoring a time delay associated with each of the one or more data packets, wherein the time delay indicates a difference between packet arrival and packet departure times. The method may also include sorting the time delay results based upon an increasing order of time delay and determining a total number of data packets associated with each of a plurality of channels. The method may also include scheduling a data packet for processing based upon, at least in part, at least one of the sorted time delay results and the total number of data packets associated with each channel.

Abstract: The present disclosure is directed towards an audio conferencing method. Some embodiments may include receiving, at a first mixing device, a first audio stream from one or more participant conferencing devices. The method may further include generating a top-N voice stream at the first mixing device, wherein the top-N voice stream corresponds with at least one top-N talker and wherein the identification of the at least one top-N talker is based upon, at least in part, an activity ranking. The method may also include receiving the top-N voice stream at a centralized mixing device and generating at least one mixed audio stream at the centralized mixing device.

Abstract: The present disclosure is directed towards an audio conferencing method. Some embodiments may include receiving, at a first mixing device, an audio signal from a first user associated with an audio conference. Embodiments may further include processing the audio signal at the first mixing device to generate a processed audio signal and transmitting the processed audio signal to a second mixing device, wherein the first mixing device and the second mixing device are distributed over a network in a cascaded configuration. Embodiments may also include receiving, at the second mixing device, a third audio signal from a second user associated with the audio conference and processing the third audio signal at the second mixing device to generate a second processed audio signal.

Abstract: A system and method configured for use in a text-to-speech (TTS) system is provided. Embodiments may include identifying, using one or more processors, a word or phrase as a named entity and identifying a language of origin associated with the named entity. Embodiments may further include transliterating the named entity to a script associated with the language of origin. If the TTS system is operating in the language of origin, embodiments may include passing the transliterated script to the TTS system. If the TTS system is not operating in the language of origin, embodiments may include generating a phoneme sequence in the language of origin using a grapheme to phoneme (G2P) converter.

Abstract: The present disclosure is directed towards a method for scheduling data packets in a multi-channel packet processing environment. The method may include receiving one or more data packets associated with an incoming signal and inserting the one or more data packets into a queue. The method may further include monitoring a time delay associated with each of the one or more data packets, wherein the time delay indicates a difference between packet arrival and packet departure times. The method may also include sorting the time delay results based upon an increasing order of time delay and determining a total number of data packets associated with each of a plurality of channels. The method may also include scheduling a data packet for processing based upon, at least in part, at least one of the sorted time delay results and the total number of data packets associated with each channel.

Abstract: The present disclosure is directed towards a method for scheduling data packets in a multi-channel packet processing environment. The method may include receiving one or more data packets associated with an incoming signal and inserting the one or more data packets into a queue. The method may further include monitoring a time delay associated with each of the one or more data packets, wherein the time delay indicates a difference between packet arrival and packet departure times. The method may also include sorting the time delay results based upon an increasing order of time delay and determining a total number of data packets associated with each of a plurality of channels. The method may also include scheduling a data packet for processing based upon, at least in part, at least one of the sorted time delay results and the total number of data packets associated with each channel.

Abstract: A system and method configured for use in a text-to-speech (TTS) system is provided. Embodiments may include identifying, using one or more processors, a word or phrase as a named entity and identifying a language of origin associated with the named entity. Embodiments may further include transliterating the named entity to a script associated with the language of origin. If the TTS system is operating in the language of origin, embodiments may include passing the transliterated script to the TTS system. If the TTS system is not operating in the language of origin, embodiments may include generating a phoneme sequence in the language of origin using a grapheme to phoneme (G2P) converter.

Abstract: The present disclosure is directed towards a method for discontinuous transmission (“DTX”) bandwidth reduction. The method may include receiving, at a processor, a frame identified as speech and determining that the frame was mistakenly identified as speech based upon, at least in part, a voice activity detection algorithm. The method may further include labeling the frame as a silence indicator frame.

Abstract: The present disclosure is directed towards a method for speech intelligibility. The method may include receiving, at one or more computing devices, a first speech input from a first user and performing voice activity detection upon the first speech input. The method may also include analyzing a spectral tilt associated with the first speech input, wherein analyzing includes computing an impulse response of a linear predictive coding (“LPC”) synthesis filter in a linear pulse code modulation (“PCM”) domain and wherein the one or more computing devices includes an adaptive high pass filter configured to recalculate one or more linear prediction coefficients.

Abstract: The present disclosure is directed towards a method for scheduling data packets in a multi-channel packet processing environment. The method may include receiving one or more data packets associated with an incoming signal and inserting the one or more data packets into a queue. The method may further include monitoring a time delay associated with each of the one or more data packets, wherein the time delay indicates a difference between packet arrival and packet departure times. The method may also include sorting the time delay results based upon an increasing order of time delay and determining a total number of data packets associated with each of a plurality of channels. The method may also include scheduling a data packet for processing based upon, at least in part, at least one of the sorted time delay results and the total number of data packets associated with each channel.

Abstract: The present disclosure is directed towards a method for discontinuous transmission (“DTX”) bandwidth reduction. The method may include receiving, at a processor, a frame identified as speech and determining that the frame was mistakenly identified as speech based upon, at least in part, a voice activity detection algorithm. The method may further include labeling the frame as a silence indicator frame.

Abstract: The present disclosure is directed towards a method for speech intelligibility. The method may include receiving, at one or more computing devices, a first speech input from a first user and performing voice activity detection upon the first speech input. The method may also include analyzing a spectral tilt associated with the first speech input, wherein analyzing includes computing an impulse response of a linear predictive coding (“LPC”) synthesis filter in a linear pulse code modulation (“PCM”) domain and wherein the one or more computing devices includes an adaptive high pass filter configured to recalculate one or more linear prediction coefficients.

Abstract: A system and method for using bi-directional conversation data to improve signal presence detection are disclosed. The detector module is adapted to communicate with a signal enhancement module. The detector module collects data from a transmit direction of the connection and a receive direction of a data connection. The collected data from the transmit and the receive direction is used to classify at least one of data in the transmit direction and data in the receive direction. Responsive to the classification, the signal enhancement module enhances data in one of the transmit direction and the receive direction. Hence, data classification accuracy is improved by using data from both the transmit and receive directions. In one embodiment, the detector module applies a voice activity detection module (VAD) process to detect the presence or absence of voice data in the collected data.

Abstract: The present disclosure is directed towards an audio conferencing method. Some embodiments may include receiving, at a first mixing device, an audio signal from a first user associated with an audio conference. Embodiments may further include processing the audio signal at the first mixing device to generate a processed audio signal and transmitting the processed audio signal to a second mixing device, wherein the first mixing device and the second mixing device are distributed over a network in a cascaded configuration. Embodiments may also include receiving, at the second mixing device, a third audio signal from a second user associated with the audio conference and processing the third audio signal at the second mixing device to generate a second processed audio signal.

Abstract: The present disclosure is directed towards an audio conferencing method. Some embodiments may include receiving, at a first mixing device, a first audio stream from one or more participant conferencing devices. The method may further include generating a top-N voice stream at the first mixing device, wherein the top-N voice stream corresponds with at least one top-N talker and wherein the identification of the at least one top-N talker is based upon, at least in part, an activity ranking. The method may also include receiving the top-N voice stream at a centralized mixing device and generating at least one mixed audio stream at the centralized mixing device.

Abstract: An automated system for implementing motivational and disciplinary approaches may include a monitoring device that determines whether a user has completed a target task, and an access control application and/or a reward management application operatively connected to the monitoring device. The reward management application may also be operatively connected to the access control application. The access control application may grant access to a desired activity upon receiving a communication from the monitoring device that the target task has been completed, while the reward management application may deposit reward points into a reward bank upon receiving a communication from the monitoring device that the target task has been completed. Alternatively, the access control application may be capable of granting access to a desired activity upon receiving a communication from the reward management application, which may also debit reward points from the reward bank when a desired activity is accessed.

Abstract: A system and method for using bi-directional conversation data to improve signal presence detection are disclosed. The detector module is adapted to communicate with a signal enhancement module. The detector module collects data from a transmit direction of the connection and a receive direction of a data connection. The collected data from the transmit and the receive direction is used to classify at least one of data in the transmit direction and data in the receive direction. Responsive to the classification, the signal enhancement module enhances data in one of the transmit direction and the receive direction. Hence, data classification accuracy is improved by using data from both the transmit and receive directions. In one embodiment, the detector module applies a voice activity detection module (VAD) process to detect the presence or absence of voice data in the collected data.

Abstract: A system and method for using bi-directional conversation data to improve signal presence detection are disclosed. The detector module is adapted to communicate with a signal enhancement module. The detector module collects data from a transmit direction of the connection and a receive direction of a data connection. The collected data from the transmit and the receive direction is used to classify at least one of data in the transmit direction and data in the receive direction. Responsive to the classification, the signal enhancement module enhances data in one of the transmit direction and the receive direction. Hence, data classification accuracy is improved by using data from both the transmit and receive directions. In one embodiment, the detector module applies a voice activity detection module (VAD) process to detect the presence or absence of voice data in the collected data.