Abstract:
This disclosure falls into the field of voice communication systems, more specifically it is related to the field of voice quality estimation in a packet based voice communication system. In particular the disclosure provides methods, computer program products and devices for reducing a prediction error of the voice quality estimation by considering forward error correction of lost voice packets.
Abstract:
A method for delivering media to a playback device including outputting first test media to be viewed by a first user. The method further includes receiving a first user input related to a first perception of the first test media by the first user and indicating a first personalized quality of experience of the first user with respect to the first test media. The method further includes generating a first personalized sensitivity profile including one or more viewing characteristics of the first user based on the first user input, and determining, based at least in part on the first personalized sensitivity profile, a first media parameter. The first media parameter is determined in order to increase an efficiency of media delivery to the first playback device over a network while preserving the first personalized quality of experience of the first user.
Abstract:
A voice quality estimation process may be triggered by receiving one or more alarms corresponding to one or more endpoint terminals being used during a teleconference. The alarm(s) may include uplink transmission alarms, downlink transmission alarms and/or acoustic quality alarms. The alarms may be based on evaluating transmission metrics and/or acoustic quality metrics. The voice quality estimation process may require a relatively greater computational burden than the processes of evaluating the transmission metrics and/or acoustic quality metrics for the purpose of potentially triggering an alarm. The accuracy and computational complexity of voice quality estimation may be adjusted by selecting times during which alarm detection will take place, alarm detector thresholds, alarm analyzer thresholds and/or levels of voice quality estimation.
Abstract:
Systems and methods are described for measuring capture performance of multiple voice signals. A first speech signal is applied to a device, and measured at a far-end of a testing environment. A second speech signal is separately applied to the device, and is also measured at the far end. The measured speech signals are added, and a quality assessment model is applied to the first far-end combined signal to obtain a first quality metric. The first speech signal and the second speech signal are then both applied at the same time to the device and measured at the far-end. The quality assessment model is applied to the second far-end combined signal to obtain a second quality metric. The quality metric for the second far-end combined signal is normalized, based on the first quality metric, to obtain a performance index for the device.
Abstract:
This disclosure falls into the field of voice communication systems, more specifically it is related to the field of voice quality estimation in a packet based voice communication system. In particular the disclosure provides a method and device for 5 reducing a prediction error of the voice quality estimation by considering the content of lost packets. Furthermore, this disclosure provides a method and device which uses a voice quality estimating algorithm to calculate the voice quality estimate based on an input which is switchable between a first and a second input mode.