The method includes splitting a radio signal received at an in-vehicle antenna into two RF streams, and at the device under test, converting the first RF stream into a demodulated audio signal and transmitting it to the tester and recorder device. The tester and recorder device also receives the demodulated audio signal, determines a spectrum of frequencies over time, inputs the spectrum of frequencies into an artificial intelligence (AI) module of audio abnormality detection. The device also receives the second RF stream, converts it into a data signal, and records the data signal into a temporary storage memory. Then, when the AI module outputs of a positive detection of audio abnormality, the device transfers data from the temporary storage memory into a permanent storage memory, where the transferred data corresponds to a time window including the detected audio abnormality.

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.

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.

A communication transmission device (1) is provided with an input sound level detection unit (11) for dividing, per unit time, sound data (100) of a predetermined period of time and converting the sound data into a bit string (101) according to whether or not a sound level thereof exceeds a predetermined threshold value, an arithmetic processing unit (10) for performing predetermined arithmetic processing on the sound data (100), an output sound level detection unit (12) for dividing, per the unit time, sound data (100a) of the predetermined period of time after the arithmetic processing and converting the sound data into a bit string (101a) according to whether or not a sound level thereof exceeds the predetermined threshold value, and a comparison determination unit (13) for determining whether or not a sound failure has occurred on the basis of a predetermined logic by which the bit string (101) before the arithmetic processing and the bit string (101a) after the arithmetic processing are compared.

A communication transmission device (1) is provided with an input sound level detection unit (11) for dividing, per unit time, sound data (100) of a predetermined period of time and converting the sound data into a bit string (101) according to whether or not a sound level thereof exceeds a predetermined threshold value, an arithmetic processing unit (10) for performing predetermined arithmetic processing on the sound data (100), an output sound level detection unit (12) for dividing, per the unit time, sound data (100a) of the predetermined period of time after the arithmetic processing and converting the sound data into a bit string (101a) according to whether or not a sound level thereof exceeds the predetermined threshold value, and a comparison determination unit (13) for determining whether or not a sound failure has occurred on the basis of a predetermined logic by which the bit string (101) before the arithmetic processing and the bit string (101a) after the arithmetic processing are compared.

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.