Abstract: A technique for suppressing non-stationary noise, such as wind noise, in an audio signal is described. In accordance with the technique, a series of frames of the audio signal is analyzed to detect whether the audio signal comprises non-stationary noise. If it is detected that the audio signal comprises non-stationary noise, a number of steps are performed. In accordance with these steps, a determination is made as to whether a frame of the audio signal comprises non-stationary noise or speech and non-stationary noise. If it is determined that the frame comprises non-stationary noise, a first filter is applied to the frame and if it is determined that the frame comprises speech and non-stationary noise, a second filter is applied to the frame.

Abstract: A method and system for improving speech quality may include estimating at least one component of a distorted portion of a speech signal from at least one component of an undistorted portion of the speech signal and reinforcing the component of the distorted portion based on the estimating. The components may include the pitch, spectral envelope and spectral energy of the speech signal. The undistorted portion of the speech signal may be delayed and the components of the distorted portion may be interpolated from the components of a delayed undistorted portion and a current undistorted portion of the speech signal. The components of the distorted portion of the speech signal may be extrapolated from a current undistorted portion of the speech signal. Components of the distorted portion of the speech signal may be estimated from frequency bands other than the frequency band affected by the distortion.

Abstract: A system and method is described for updating the state of an audio decoder, such as a CVSD decoder, after a packet loss has occurred. In response to the loss of a packet, the system and method encodes audio samples produced by a packet loss concealment (PLC) algorithm and effectively passes the encoded audio samples through the audio decoder in lieu of the contents of the lost packet. This operation brings the state of the audio decoder into better synchronization with the state of a remote audio encoder, thereby reducing or minimizing the degrading effect of the packet loss on the perceived quality of an output audio signal produced by a voice processing system that includes the audio decoder.

Abstract: Various embodiments are disclosed relating to the real-time monitoring and control for audio devices. An apparatus may include a peripheral audio device configured to operate in an operational mode or a debug mode, the peripheral audio device including an audio enhancement logic configured to include at least one tunable parameter. The apparatus may also include the peripheral audio device being further configured to transmit and receive data via a data channel to allow a debug or test to be performed on the peripheral audio device, while operating in the debug mode, and the at least one tunable parameter to be adjusted.

Abstract: A technique for suppressing non-stationary noise, such as wind noise, in an audio signal is described. In accordance with the technique, a series of frames of the audio signal is analyzed to detect whether the audio signal comprises non-stationary noise. If it is detected that the audio signal comprises non-stationary noise, a number of steps are performed. In accordance with these steps, a determination is made as to whether a frame of the audio signal comprises non-stationary noise or speech and non-stationary noise. If it is determined that the frame comprises non-stationary noise, a first filter is applied to the frame and if it is determined that the frame comprises speech and non-stationary noise, a second filter is applied to the frame.

Abstract: A method and system for improving speech quality may include estimating at least one component of a distorted portion of a speech signal from at least one component of an undistorted portion of the speech signal and reinforcing the component of the distorted portion based on the estimating. The components may include the pitch, spectral envelope and spectral energy of the speech signal. The undistorted portion of the speech signal may be delayed and the components of the distorted portion may be interpolated from the components of a delayed undistorted portion and a current undistorted portion of the speech signal. The components of the distorted portion of the speech signal may be extrapolated from a current undistorted portion of the speech signal. Components of the distorted portion of the speech signal may be estimated from frequency bands other than the frequency band affected by the distortion.

Abstract: Various embodiments are disclosed relating to the real-time monitoring and control for audio devices. An apparatus may include a peripheral audio device configured to operate in an operational mode or a debug mode, the peripheral audio device including an audio enhancement logic configured to include at least one tunable parameter.