Concealing a lost audio frame of a received audio signal is provided by performing a sinusoidal analysis (81) of a part of a previously received or reconstructed audio signal, wherein the sinusoidal analysis involves identifying frequencies of sinusoidal components of the audio signal, applying a sinusoidal model on a segment of the previously received or reconstructed audio signal, wherein said segment is used as a prototype frame in order to create a substitution frame for a lost audio frame, and creating the substitution frame (83) for the lost audio frame by time-evolving sinusoidal components of the prototype frame, up to the time instance of the lost audio frame, in response to the corresponding identified frequencies.

Concealing a lost audio frame of a received audio signal is provided by performing a sinusoidal analysis (81) of a part of a previously received or reconstructed audio signal, wherein the sinusoidal analysis involves identifying frequencies of sinusoidal components of the audio signal, applying a sinusoidal model on a segment of the previously received or reconstructed audio signal, wherein said segment is used as a prototype frame in order to create a substitution frame for a lost audio frame, and creating the substitution frame (83) for the lost audio frame by time-evolving sinusoidal components of the prototype frame, up to the time instance of the lost audio frame, in response to the corresponding identified frequencies.

A voice quality estimation apparatus according to one embodiment includes: first sequence creation means for creating a first sequence by applying a first characteristic indicating that quality degradation caused by packet loss is perceived by a user all at once, to a sequence consisting of elements each indicating whether or not a packet of a voice call has been lost; second sequence creation means for creating a second sequence by applying a second characteristic indicating that the larger the quality degradation is, the more likely the user is to perceive the quality degradation, to the first sequence created by the first sequence creation means; third sequence creation means for creating a third sequence by applying a third characteristic indicating that packet loss concealment alleviates the quality degradation to be perceived, to the second sequence created by the second sequence creation means; calculation means for calculating a degradation amount per unit time from the third sequence created by the third sequence creation means; and estimation means for estimating voice quality that is to be experienced by the user, from the degradation amount calculated by the calculation means, using a mapping function that indicates a relationship between the degradation amount regarding the voice quality and a voice quality evaluation value that is based on the user's subjectivity.

A voice quality estimation apparatus according to one embodiment includes: first sequence creation means for creating a first sequence by applying a first characteristic indicating that quality degradation caused by packet loss is perceived by a user all at once, to a sequence consisting of elements each indicating whether or not a packet of a voice call has been lost; second sequence creation means for creating a second sequence by applying a second characteristic indicating that the larger the quality degradation is, the more likely the user is to perceive the quality degradation, to the first sequence created by the first sequence creation means; third sequence creation means for creating a third sequence by applying a third characteristic indicating that packet loss concealment alleviates the quality degradation to be perceived, to the second sequence created by the second sequence creation means; calculation means for calculating a degradation amount per unit time from the third sequence created by the third sequence creation means; and estimation means for estimating voice quality that is to be experienced by the user, from the degradation amount calculated by the calculation means, using a mapping function that indicates a relationship between the degradation amount regarding the voice quality and a voice quality evaluation value that is based on the user's subjectivity.

Transmitting node and receiving node for audio coding and methods therein. The nodes being operable to encode/decode speech and to apply a discontinuous transmission (DTX) scheme comprising transmission/reception of Silence Insertion Descriptor (SID) frames during speech inactivity. The method in the transmitting node comprising determining, from amongst a number N of hangover frames, a set Y of frames being representative of background noise, and further transmitting the N hangover frames, comprising at least said set Y of frames, to the receiving node. The method further comprises transmitting a first SID frame to the receiving node in association with the transmission of the N hangover frames, where the SID frame comprises information indicating the determined set Y of hangover frames to the receiving node. The method enables the receiving node to generate comfort noise based on the hangover frames most adequate for the purpose.

Transmitting node and receiving node for audio coding and methods therein. The nodes being operable to encode/decode speech and to apply a discontinuous transmission (DTX) scheme comprising transmission/reception of Silence Insertion Descriptor (SID) frames during speech inactivity. The method in the transmitting node comprising determining, from amongst a number N of hangover frames, a set Y of frames being representative of background noise, and further transmitting the N hangover frames, comprising at least said set Y of frames, to the receiving node. The method further comprises transmitting a first SID frame to the receiving node in association with the transmission of the N hangover frames, where the SID frame comprises information indicating the determined set Y of hangover frames to the receiving node. The method enables the receiving node to generate comfort noise based on the hangover frames most adequate for the purpose.

A method for decoding a digital signal encoded using predictive coding and transform coding, comprising the following steps: predictive decoding of a preceding frame of the digital signal, encoded by a set of predictive coding parameters; detecting the loss of a current frame of the encoded digital signal; generating by prediction, from at least one predictive coding parameter encoding the preceding frame, a frame for replacing the current frame; generating by prediction, from at least one predictive coding parameter encoding the preceding frame, an additional segment of digital signal; temporarily storing said additional segment of digital signal.

A method for decoding a digital signal encoded using predictive coding and transform coding, comprising the following steps: predictive decoding of a preceding frame of the digital signal, encoded by a set of predictive coding parameters; detecting the loss of a current frame of the encoded digital signal; generating by prediction, from at least one predictive coding parameter encoding the preceding frame, a frame for replacing the current frame; generating by prediction, from at least one predictive coding parameter encoding the preceding frame, an additional segment of digital signal; temporarily storing said additional segment of digital signal.

A method for generating voice in an ongoing call session based on artificial intelligent techniques is provided. The method includes extracting a plurality of features from a voice input through an artificial neural network (ANN); identifying one or more lost audio frames within the voice input; predicting by the ANN, for each of the one or more lost audio frames, one or more features of the respective lost audio frame; and superposing the predicted features upon the voice input to generate an updated voice input.

The present document discloses a method for fading discontinued audio feeds for replay by a speaker. In particular, the method may first comprise receiving an input audio feed comprising a plurality of samples. The method may further comprise determining whether the input audio feed is discontinued. And, when discontinuity of the input audio feed is detected, the method may comprise generating an intermediate audio signal comprising a plurality of samples based on the discontinued input audio feed. In particular, the intermediate audio signal may be generated based on a last portion of the discontinued input audio feed that has been output for replay. In addition, the method may further comprise applying a fadeout function to the intermediate audio signal to generate a fadeout audio signal. Finally, the method may comprise outputting the fadeout audio signal for replay by the speaker.