Dialpad, Inc.

Nuance Communications, Inc.

Qian-Yu Tang

John Rector

Victor Zeyliger

Sridhar Pilli

Weiying Li

Mahesh Godavarti

Jose Lainez

Jagadeesh Balam

Kathiravan Ramaswami

Franck Bonard

Corey Burke

Constantine V Skidanenko

A system and method are described for automatic acoustic feedback cancellation in real time. In some implementations, the system may receive audio data describing an audio signal, which the system may use to determine a set of frames of the audio signal. Spectral analysis may be performed on the one or more frames of the audio to detect spectral patterns of two or more frames indicative of acoustic feedback. An additional delay identification test may be performed to identify a consistent delay indicative of acoustic feedback. In some implementations, a state machine is advanced based in part on accumulated delay votes. Decisions can be made to mute the acoustic feedback and cease the muting operation when silence is detected.

System and method for smart feedback cancellation

A system and method are described for smart noise cancellation. In some implementations, the system may receive audio data describing an audio signal, which the system may use to determine a set of frames of the audio signal. Spectral analysis, which may include a signal-to-noise ratio estimate, may be performed on the one or more frames of the audio. In some instances, the system may identify a noise frame from among the one or more frames based on the spectral analysis, and may reduce noise in the one or more frames based on the noise frame and the spectral analysis on the one or more frames of the audio signal.

Smart noise cancellation

Audio data describing an audio signal may be received and used to determine a set of frames of the audio signal. One or more potential music events may be determined in the audio signal using a spectral analysis of the set of frames. The audio signal may be analyzed for one or more potential noise or tone events. One or more music states of the audio signal may be determined based on the one or more potential music events and a presence or absence of the one or more noise or tone events. Audio enhancement of the audio signal may be modified based on the one or more determined states of the audio signal.

Chroma detection among music, speech, and noise

Audio data describing an audio signal may be received and used to determine a set of frames of the audio signal. A plurality of note onsets in the set of frames may be identified based on spectral energy of the audio signal in the set of frames. One or more tempos may be computed based on the identified plurality of note onsets. The one or more tempos may be validated based on a tempo validation condition. One or more music states of the audio signal may be determined based on the validated one or more tempos. Audio enhancement of the audio signal may be modified based on the one or more determined states of the audio signal.

Smart voice enhancement architecture for tempo tracking among music, speech, and noise

The present disclosure is directed towards a system and method for reducing tandeming effects in a communications system. The method may include receiving, at a speech decoder, an input bitstream associated with an incoming initial speech signal from a speech encoder. The method may further include determining whether or not coding is required and if coding is required, modifying an excitation signal associated with the bitstream. The method may also include providing the modified excitation signal to an adaptive encoder.

System and method for reducing tandeming effects in a communication system

The present disclosure is directed towards a system and method for handling rogue data packets. The method may include receiving, using one or more processors, a first data packet having header information associated therewith. The method may further include obtaining, from the header information, sequence number, timestamp and synchronization source identifier information. The method may also include detecting one or more rogue data packets, based upon, at least in part, at least one of the sequence number, timestamp and synchronization source identifier information.

System and method for handling rogue data packets

Embodiments included herein are directed towards a system and method for addressing discontinuous transmission (DTX) in a network device. Embodiments may include receiving, at a computing device, an audio signal and generating at least one silence descriptor (SID) frame associated with the audio signal. Embodiments may also include generating at least one no data frame associated with the audio signal. Embodiments may also include initiating a speech decoder, voice enhancement, and speech encoder operation for the at least one SID frame during a DTX operation and bypassing the speech decoder, voice enhancement, and speech encoder functions for the at least one no data frame.

System and method for addressing discontinuous transmission in a network device

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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Milpitas, CA, United States of America

Qian-Yu Tang