Method for processing acoustic frequency flow playback in network terminal buffer

Abstract

The invention discloses a method for processing audio stream playback in the network terminal buffer zone for solving the problem of voice pausing and jamming in the network communication. The aim of the invention is achieved by performing real time audio (e.g. voice) communication on the packet-switching network (e.g. IP network), arranging a shake buffer zone on the receiving end, after the receiving end receives the audio package, it first performs decoding based on the normal sequence, then places it into the shake buffer zone, when the shake buffer zone is to be filled, lower the sampling rate to the audio data to realize the fast playback of the audio data stream, when the shake buffer zone is to be empty, raise the sampling rate to the audio data to realize the low speed playback of the audio data stream, when the audio data in the shake buffer is within the normal range, playback the audio stream with the original sampling rate.

Description

technical field [0001] The invention relates to a method for processing audio streams, in particular to a method for processing audio stream playback in a network terminal buffer for continuous, smooth and normal playback of calls on a packet switching network. Background technique [0002] At present, due to the rapid development of packet-switching networks, voice real-time communications originally carried on circuit-switching networks have begun to be transferred to IP packet-switching networks in the form of packet-switched voice (VoIP) on a large scale, eventually forming audio and data and Other streaming converged networks. There are some different characteristics between the traditional circuit switching network and the packet switching network. For example, the circuit switching network provides end-to-end fixed bandwidth and exclusive communication lines, and there will be no packet loss, disorder, delay jitter, etc. that exist on the packet switching network. Pr...

AI Technical Summary

Problems solved by technology

The larger the jitter buffer, the greater the jitter that can be smoothed, but an excessively large jitter buffer will cause an increase in audio playback delay, and an excessively large delay is also undesirable and will cause difficulties in real-time communication; jitter The size of the buffer can be set according to specific needs, and can also be dynamically adjusted according to network conditions

But this does not effectively guarantee the continuous playback of the audio stream

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