Method and system for maintaining real-time audio stream play delay in reliable transport network

Abstract

The invention discloses a method and a system for maintaining real-time audio stream playing delay in a reliable transmission network. The method comprises the following steps of: calculating receiving time of an audio packet; detecting whether the actual receiving time of the audio packet is earlier than the calculated receiving time, if so, defining the minimum delay audio packet, and if not, counting the detection times; determining whether the number of times of detection is equal to a preset number of times, if not, returning to the first step, and if yes, defining the minimum delay audio packet, and calculating the playing start time of each audio packet based on the receiving time of the minimum delay audio packet, the system fixed delay time and the time length of the audio packet; and playing the audio packet when judging that the playing start time of the audio packet does not exceed the preset playing start time.

Description

technical field [0001] The present invention relates to audio stream playback, in particular to a method and system for maintaining real-time audio stream playback delay in a reliable transmission network. Background technique [0002] In recent years, Bluetooth headsets are used to connect mobile phones or TVs to watch videos and play games, and the issue of audio and video synchronization is paid special attention to. [0003] The continuous audio data stream is cut into blocks and compressed at the sending end (mobile phone or TV), and then composed of audio packets sent to the receiving end (headphones) through a wired or wireless network, and then the audio packets are reassembled and decompressed at the receiving end. After compression, it is restored to continuous audio data and played out. [0004] In the process of network transmission, due to various reasons such as network congestion, interference and lost retransmission, the delay time of each audio packet from ...

AI Technical Summary

Problems solved by technology

When the network congestion has just been relieved, a large number of audio packets accumulated in the transmission queue of the sending end may cause a transmission burst, which will quickly fill up the jitter buffer of the receiving end and exceed a lot, and the playback delay will be reduced in this way. Significantly exceeds the scheduled 150ms

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