There is provided mechanisms for handling poor audio quality. A method is performed by a receiving terminal device. The method comprises obtaining an indication of poor audio quality of incoming audio at the receiving terminal device. The incoming audio originates from a transmitting terminal device. The method comprises initiating text conversion of the incoming audio. The method comprises receiving text resulting from automatic speech recognition having been applied to the incoming audio. The method comprises providing a representation of the text to a user interface of the receiving terminal device.

An information processing apparatus includes: a processor configured to instantaneously acquire quality information indicative of quality of utterer's voice on a listener's side; and instantaneously present improvement information for improving the quality to the utterer in a case where the quality indicated by the acquired quality information does not satisfy a predetermined condition.

The present disclosure provides methods and systems that may be used for providing quality control for audio samples. The audio samples may be speech samples of a user. The user may be participating in an audio interview.

A method of determining the quality of transmitted voice data can include: providing voice data at a transmitter side in a first data format, providing a first test signal in the first data format, combining the voice data and the test signal to form input data, transmitting the input data in a transmittal data format, receiving the transmitted input data at a receiver side to obtain output data, removing at least portions of a data packet in the output data or of a data packet derived therefrom in order to derive a second test signal, and analysing the derived second test signal by applying a predetermined analysis criterion in order to obtain at least one value for a quality indicator. An apparatus and system can also be configured to utilize embodiments of the method.

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.

A new audio quality assessment system includes an assessment system running in a receiver system of a VoIP communication system. The new audio quality assessment system determines an accurate MOS of a VoIP call within a time window. The audio quality assessment system determines an effective PLC counter, a PLC impact factor, an effective AS counter, an AS impact factor, a network impact factor, a codec type of the received voice packets, a bitrate of the received voice packets, an initial MOS from a configured codec-bitrate MOS table, and determines the accurate MOS based on these data. The determined MOS is more accurate and efficiently obtained since it is based on efficiently collected statistics of the receiver system's modules and a pre-configured codec-bitrate MOS table.

Methods and systems are disclosed herein for improving the quality of audio for use in a biometric. A biometric system may use machine learning to determine whether audio or a portion of the audio should be used as a biometric for a user. A sample of the user's voice may be used to generate a voice signature of the user. Portions of the audio that do not meet a similarity threshold when compared with the voice signature may be removed from the audio. Additionally or alternatively, interfering noises may be detected and removed from the audio to improve the quality of a voice biometric generated from the audio.

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 feedback control system for data transmissions in voice activated data packet based computer network environment is provided. A system can receive audio signals detected by a microphone of a device. The system can parse the audio signal to identify trigger keyword and request. The system can select a content item using the trigger keyword or request. The content item can be configured to establish a communication session between the device and a third party device. The system can monitor the communication session to measure a characteristic of the communication session. The system can generate a quality signal based on the measured characteristic.

A conferencing system terminal device includes a display, an audio output, a user interface, a communication device, and one or more processors. The one or more processors present an audio difficulties user actuation target upon the display during an audio or video conference occurring across a network and concurrently with a presentation of conference content. Actuation of the audio difficulties user actuation target indicates that audio content associated with the audio or video conference being delivered by the audio output is impaired.