Abstract: There is provided a method of post-processing a speech signal. The method comprises applying a time-domain post-processing to the speech signal, using LPC coefficients, for a low-band frequency range and applying a frequency-domain post-processing to the speech signal, using MDCT coefficients, for the high-band frequency range. Applying the frequency-domain post-processing includes decoding an encoded speech signal to obtain MDCT coefficients representative of the speech signal divided into a plurality of sub-bands, generating an envelope for each sub-band of the plurality of sub-bands as an average magnitude of the MDCT coefficients of the sub-band, generating an envelope modification factor for each sub-band of the plurality of sub-band using the MDCT coefficients of the sub-band, modifying the envelope by the envelope modification factor for each sub-band of the plurality of sub-bands to provide a modified envelope, and generating the post-processed speech signal using the modified envelope.

Abstract: The present invention relates to a method and arrangement for improving quality of a voice transmission by extracting filter coefficient parameters with respect to a voice signal in a first speech transmission rate, and using the extracted filter coefficient parameters in a second transmission rate that is equal or lower than the first transmission rate.

Abstract: Method for removing noise from a digital speech waveform, including receiving the digital speech waveform having the noise contained therein, segmenting the digital speech waveform into one or more frames, each frame having a clean portion and a noisy portion, extracting a feature component from each frame, creating an nonlinear speech distortion model from the feature components, creating a statistical noise model by making a Piecewise Linear Approximation (PLA) of the nonlinear speech distortion model, determining the clean portion of each frame using the statistical noise model, a log power spectra of each frame, and a model of a digital speech waveform recorded in a noise controlled environment, and constructing a clean digital speech waveform from each clean portion of each frame.

Abstract: A signal enhancement system improves the understandability of speech or other audio signals. The system reinforces selected parts of the signal, may attenuate selected parts of the signal, and may increase SNR. The system includes delay logic, an adaptive filter, and signal reinforcement logic. The adaptive filter may track one or more fundamental frequencies in the input signal and outputs a filtered signal. The filtered signal may approximately reproduce the input signal approximately delayed by an integer multiple of the signal's fundamental frequencies. The reinforcement logic combines the input signal and the filtered signal output to produce an enhanced signal output.