A conference terminal and an echo cancellation method for a conference are provided. In the echo cancellation method, a synthetic speech signal is received. The synthetic speech signal includes a user speech signal of a speaking party corresponding to a first conference terminal of multiple conference terminals and an audio watermark signal corresponding to the first conference terminal. One or more delay times corresponding to the audio watermark signal are detected in a received audio signal. The received audio signal is recorded through a sound receiver of a second conference terminal of the conference terminals. An echo in the received audio signal is canceled according to the delay time.

A method for Parametric resynthesis (PR) producing an audible signal. A degraded audio signal is received which includes a distorted target audio signal. A prediction model predicts parameters of the audible signal from the degraded signal. The prediction model was trained to minimize a loss function between the target audio signal and the predicted audible signal. The predicted parameters are provided to a waveform generator which synthesizes the audible signal.

A method for Parametric resynthesis (PR) producing an audible signal. A degraded audio signal is received which includes a distorted target audio signal. A prediction model predicts parameters of the audible signal from the degraded signal. The prediction model was trained to minimize a loss function between the target audio signal and the predicted audible signal. The predicted parameters are provided to a waveform generator which synthesizes the audible signal.

A method and apparatus for processing a signal. An implementation of the method includes: acquiring a reference signal of a to-be-tested voice, the reference signal being a signal output to a voice output device, where the voice output device outputs the to-be-tested voice after obtaining the reference signal; receiving, from a voice input device, an echo signal of the to-be-tested voice, the echo signal being a signal of the to-be-tested voice collected by the voice input device; performing signal preprocessing on the reference signal and the echo signal respectively; and inputting the processed reference signal and the processed echo signal into a pre-trained time delay estimation model, to obtain a time difference between the reference signal and the echo signal output by the time delay estimation model.

A noise suppression method includes transforming a time-domain input signal into an input spectrum that is the spectrum of the input signal, the input signal comprising speech components and noise components, and the input spectrum comprising a speech spectrum that is the spectrum of the speech components and a noise spectrum that is the spectrum of the noise components, smoothing magnitudes of the input spectrum to provide a smoothed-magnitude input spectrum, and estimating basic suppression filter coefficients from the input spectrum and the smoothed input spectrum. The method further includes determining noise suppression filter coefficients from the estimated basic suppression filter coefficients and a spectral correlation factor, the spectral correlation factor indicating whether speech is present in the input signal or not, filtering the input spectrum based on the noise suppression filter coefficients to generate an output spectrum; and transforming the output spectrum into a time-domain output signal.

Disclosed embodiments provide techniques for improved call quality during telephony sessions. The speech quality of an active voice session is periodically evaluated using multiple noise reduction algorithms. In an instance where the speech quality of the currently used noise reduction algorithm is below the quality of another noise reduction algorithm, the telephony system may switch to a new noise reduction algorithm as the currently used (active) noise reduction algorithm in order to improve call quality during an active voice session.

Disclosed embodiments provide techniques for improved call quality during telephony sessions. The speech quality of an active voice session is periodically evaluated using multiple noise reduction algorithms. In an instance where the speech quality of the currently used noise reduction algorithm is below the quality of another noise reduction algorithm, the telephony system may switch to a new noise reduction algorithm as the currently used (active) noise reduction algorithm in order to improve call quality during an active voice session.

Communication terminal includes a first microphone system, a second microphone system, and a noise reduction processing unit (NRPU). The NRPU receives a primary signal from the first microphone system and a secondary signal from the second microphone system. The NRPU dynamically identify an optimal transfer function of a correction filter which can be applied to the secondary signal provided by the second microphone system to obtain a correction signal. The correction signal is subtracted from the primary signal to obtain a remainder signal which approximates a signal of interest contained within the primary signal.

Communication terminal includes a first microphone system, a second microphone system, and a noise reduction processing unit (NRPU). The NRPU receives a primary signal from the first microphone system and a secondary signal from the second microphone system. The NRPU dynamically identify an optimal transfer function of a correction filter which can be applied to the secondary signal provided by the second microphone system to obtain a correction signal. The correction signal is subtracted from the primary signal to obtain a remainder signal which approximates a signal of interest contained within the primary signal.

Communication terminal includes a first microphone system, a second microphone system, and a noise reduction processing unit (NRPU). The NRPU receives a primary signal from the first microphone system and a secondary signal from the second microphone system. The NRPU dynamically identify an optimal transfer function of a correction filter which can be applied to the secondary signal provided by the second microphone system to obtain a correction signal. The correction signal is subtracted from the primary signal to obtain a remainder signal which approximates a signal of interest contained within the primary signal.