Transmissão de áudio com baixa latência sobre redes wireless IEEE 802.11 com foco em acessibilidade para o cinema digital

Detalhes bibliográficos
Ano de defesa: 2021
Autor(a) principal: Guedes, Caio Marcelo Campoy
Orientador(a): Não Informado pela instituição
Banca de defesa: Não Informado pela instituição
Tipo de documento: Dissertação
Tipo de acesso: Acesso aberto
Idioma: por
Instituição de defesa: Universidade Federal da Paraíba
Brasil
Informática
Programa de Pós-Graduação em Informática
UFPB
Programa de Pós-Graduação: Não Informado pela instituição
Departamento: Não Informado pela instituição
País: Não Informado pela instituição
Palavras-chave em Português:
FEC
Link de acesso: https://repositorio.ufpb.br/jspui/handle/123456789/20982
Resumo: The transmission of audio reinforcement from a server to mobile phones used by the hearing impaired during the presentation of a film in cinemas is an example among applications that use Wi-Fi networks and require the transmission of audio with low latency, synchronism and resilience. In this application, it is meant that the delay between capturing and repro- ducing the audio stream lays below 40 ms, a requirement that focus on the optimization of encoding, transmission and decoding of the solution. This study aims to develop a client-server solution capable of transmitting audio in real time over IEEE 802.11 wireless networks between a server and clients running on mobile devices. The work seeks to limit the maximum latency between source and player to a threshold of 40 milliseconds defined as a threshold for the perception of lip sync. Maintaining low latency by avoiding variations in delay and packet loss in volatile environments such as wireless networks is a complex problem. Therefore, aiming for maximum resilience, the solution makes use of Forward Error Correction and proposes the use of temporal redundancy to mitigate the undesirable effects caused by the loss and discarding of delayed packages. Additionally, a new adaptive algorithm is proposed and used to adjust the buffer of playback in order to mitigate the network jitter.