Nuance Communications, Inc.

Mahesh Godavarti

Sridhar Pilli

Biswaranjan Panigrahi

Jose Lainez

Dushyant Sharma

Patrick A Naylor

Qian-Yu Tang

Daniel Almendro Barreda

Yifang Xu

Jagadeesh Balam

Ivan A Sanchez Quijas

Anurag Ratan Singh

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.

System and method for packet scheduling

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.

System and method for audio conferencing

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.

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.

Process for improving pronunciation of proper nouns foreign to a target language text-to-speech system

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.

System and method to reduce transmission bandwidth via improved discontinuous transmission

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.

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Cupertino, CA, United States of America

Mahesh Godavarti