Êtes-vous un étudiant de l'EPFL à la recherche d'un projet de semestre?
Travaillez avec nous sur des projets en science des données et en visualisation, et déployez votre projet sous forme d'application sur GraphSearch.
Concept
Latency (audio)
Résumé
Latency refers to a short period of delay (usually measured in milliseconds) between when an audio signal enters a system and when it emerges. Potential contributors to latency in an audio system include analog-to-digital conversion, buffering, digital signal processing, transmission time, digital-to-analog conversion and the speed of sound in the transmission medium.
Latency can be a critical performance metric in professional audio including sound reinforcement systems, foldback systems (especially those using in-ear monitors) live radio and television. Excessive audio latency has the potential to degrade call quality in telecommunications applications. Low latency audio in computers is important for interactivity.
In all systems, latency can be said to consist of three elements: codec delay, playout delay and network delay.
Latency in telephone calls is sometimes referred to as delay; the telecommunications industry also uses the term quality of experience (QoE). Voice quality is measured according to the ITU model; measurable quality of a call degrades rapidly where the mouth-to-ear delay latency exceeds 200 milliseconds. The mean opinion score (MOS) is also comparable in a near-linear fashion with the ITU's quality scale - defined in standards G.107, G.108 and G.109 - with a quality factor R ranging from 0 to 100. An MOS of 4 ('Good') would have an R score of 80 or above; to achieve 100R requires an MOS exceeding 4.5.
The ITU and 3GPP groups end-user services into classes based on latency sensitivity:
Similarly, the G.114 recommendation regarding mouth-to-ear delay indicates that most users are "very satisfied" as long as latency does not exceed 200 ms, with an according R of 90+. Codec choice also plays an important role; the highest quality (and highest bandwidth) codecs like G.711 are usually configured to incur the least encode-decode latency, so on a network with sufficient throughput sub-100 ms latencies can be achieved. G.711 at a bitrate of 64 kbit/s is the encoding method predominantly used on the public switched telephone network.
Source officielle
À propos de ce résultat
Cette page est générée automatiquement et peut contenir des informations qui ne sont pas correctes, complètes, à jour ou pertinentes par rapport à votre recherche. Il en va de même pour toutes les autres pages de ce site. Veillez à vérifier les informations auprès des sources officielles de l'EPFL.