Conexant Systems, Inc.

Verizon Patent and Licensing Inc.

Plantronics, Inc.

Youhong Lu

Trausti Thormundsson

Ragnar Hlynur Jonsson

Govind Kannan

Andrew Walters

Yair Kerner

Ravi Kalluri

Luigi Bojan

Sverrir Olafsson

Yadong Wang

Arun Rajasekaran

Vilhjalmur S Thorvaldsson

Vasanthi Vuppuluri

Chris X Gao

Pre-processing systems, methods of pre-processing, and speech processing systems for improved Automated Speech Recognition are provided. Some pre-processing systems for improved speech recognition of a speech signal are provided, which systems comprise a pitch estimation circuit; and a pitch equalization processor. The pitch estimation circuit is configured to receive the speech signal to determine a pitch index of the speech signal, and the pitch equalization processor is configured to receive the speech signal and pitch information, to equalize a speech pitch of the speech signal using the pitch information, and to provide a pitch-equalized speech signal.

Systems and methods of pre-processing of speech signals for improved speech recognition

A method, a device, and a non-transitory storage medium are described in which a power of late reverberation of a speech signal is estimated based on early samples of the speech signal. The power of the late reverberation may be subtracted linearly or non-linearly from the speech signal.

Audio processing for speech

Techniques, described herein, may enable a computing device (e.g., a set-top-box, a user equipment (UE), etc.) with an array of sensors (e.g., microphones, antennas, etc.) to engage in beamforming in a manner that optimizes the signal strength of the target signal. For example, the computing device may create an optimized beam pattern by applying a pre-selected band width to a steering direction of the target signal. The beam width may include steering vectors at equal, angular distances from the steering direction, enabling steep transition slopes to be applied to the optimized beam pattern. In some implementations, the beam width and transition slopes may be standardized, such that the optimized beam pattern for any signal may be uniform, regardless of variables such as frequency and time.

Beamforming optimization for receiving audio signals

Presented is a method and associated system for suppression of linear and nonlinear echo. The method includes dividing an input signal into several frequency bands in each of a several of time frames. The input signal may include an echo signal. The method further includes multiplying the input signal in each of the several frequency bands by a corresponding echo suppression signal. Calculating the corresponding echo suppression signal may include estimating a power of the echo signal in a particular frequency band as a sum of several component echo powers, each of the several component echo powers due to an excitation from a far-end signal in a corresponding one of the several frequency bands. Calculating the corresponding echo suppression signal may further include subtracting the power of the echo signal in the particular frequency band from a power of the input signal in the particular frequency band.

Nonlinear echo suppression

A system for detecting motion comprising a first speaker, a first microphone separated from the first speaker by a distance D, a sound generator, an echo parameter measurement device and an echo parameter monitor, wherein the echo parameter monitor stores two or more sequential echo parameters and generates a motion indicator if the two or more sequential echo parameters indicate a change in an acoustic echo path that exceeds a predetermined threshold.

Integrated motion detection using changes in acoustic echo path

Traditionally, echo cancellation has employed linear adaptive filters to cancel echoes in a two way communication system. The rate of adaptation is often dynamic and varies over time. Disclosed are novel rates of adaptation that perform well in the presence of background noise, during double talk and with echo path changes. Additionally, the echo or residual echo can further be suppressed with non-linear processing performed using joint frequency-time domain processing.

Systems and methods for echo cancellation and echo suppression

A method for continuously estimating reverberation decay comprising receiving a sequence of audio data samples. Determining whether a plateau is present in the sequence of audio data samples. Generating one or more reverberation parameters from the sequence of audio data samples if it is determined that the plateau is present.

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Irvine, CA, United States of America

Youhong Lu