An audio signal processing device comprises a discontinuity detector configured to determine an occurrence of a discontinuity from a sudden increase of an amplitude of decoded audio obtained by decoding the first audio packet which is received correctly after an occurrence of a packet loss, and a discontinuity corrector for correcting the discontinuity of the decoded audio.

A method for processing speech signals prior to encoding a digital signal comprising audio data includes selecting frequency domain coding or time domain coding based on a coding bit rate to be used for coding the digital signal and a short pitch lag detection of the digital signal.

An audio encoder for providing an encoded representation on the basis of an audio signal, wherein the audio encoder is configured to obtain a noise information describing a noise included in the audio signal, and wherein the audio encoder is configured to adaptively encode the audio signal in dependence on the noise information, such that encoding accuracy is higher for parts of the audio signal that are less affected by the noise included in the audio signal than for parts of the audio signal that are more affected by the noise included in the audio signal.

A method for processing speech signals prior to encoding a digital signal comprising audio data includes selecting frequency domain coding or time domain coding based on a coding bit rate to be used for coding the digital signal and a short pitch lag detection of the digital signal.

An apparatus for selecting one of a first encoding algorithm and a second encoding algorithm includes a filter configured to receive the audio signal, to reduce the amplitude of harmonics in the audio signal and to output a filtered version of the audio signal. First and second estimators are provided for estimating first and second quality measures in the form of SNRs of segmented SNRs associated with the first and second encoding algorithms without actually encoding and decoding the portion of the audio signal using the first and second encoding algorithms. A controller is provided for selecting the first encoding algorithm or the second encoding algorithm based on a comparison between the first quality measure and the second quality measure.

An apparatus includes a first calculator configured to determine a long-term noise estimate of the audio signal. The apparatus also includes a second calculator configured to determine a formant-sharpening factor based on the determined long-term noise estimate. The apparatus includes a filter configured to filter a codebook vector to generate a filtered codebook vector. The filter is based on the determined formant-sharpening factor, and the codebook vector is based on information from the audio signal. The apparatus further includes an audio coder configured to generate a formant-sharpened low-band excitation signal based on the filtered codebook vector.

A method for processing speech signals prior to encoding a digital signal comprising audio data includes selecting frequency domain coding or time domain coding based on a coding bit rate to be used for coding the digital signal and a short pitch lag detection of the digital signal.

A method for processing speech signals prior to encoding a digital signal comprising audio data includes selecting frequency domain coding or time domain coding based on a coding bit rate to be used for coding the digital signal and a short pitch lag detection of the digital signal.

This application relates to an audio signal enhancement method, performed by a computer device. The method including decoding received speech packets sequentially to obtain a residual signal, long term filtering parameters and linear filtering parameters; filtering the residual signal to obtain an audio signal; extracting feature parameters from the audio signal, when the audio signal is a feedforward error correction frame signal; converting the audio signal into a filter speech excitation signal based on the linear filtering parameters; performing speech enhancement on the filter speech excitation signal according to the feature parameters, the long term filtering parameters and the linear filtering parameters to obtain an enhanced speech excitation signal; and performing speech synthesis to obtain an enhanced speech signal based on the enhanced speech excitation signal and the linear filtering parameters.