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Voice data coding and decoding method

A codec method and voice data technology, which is applied in the field of Internet telephony, can solve problems such as the difficulty in achieving a balance between packet loss recovery effect and coding efficiency, and achieve good quality, repair network packet loss, and guarantee the effect of packet error recovery rate

Active Publication Date: 2017-04-19
SHENZHEN GRANDSTREAM NETWORKS TECH
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Problems solved by technology

However, the current FEC coding design has the difficulty of achieving a balance between packet loss recovery effect and coding efficiency

Method used

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  • Voice data coding and decoding method

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Embodiment Construction

[0020] The basic principle of the invention: a method for efficient FEC coding of voice data at the channel sending end, which adds a FEC redundant data to each frame of voice data and sends it through the channel. In this way, it is possible to reduce network jitter and delay as much as possible and restore voice data with the greatest probability in a poor network environment, thereby improving the voice quality of the receiving end. The specific steps of the method are from carrying out high-efficiency FEC encoding according to the voice information data and packing the streaming media packets according to the standard format and then sending them through the channel to receiving the streaming media packets containing audio data and FEC redundant data from the channel and performing verification and error correction.

[0021] In order to illustrate the technical solutions of the examples of the present invention more clearly, the implementation of the present invention will ...

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Abstract

The invention mainly discloses a voice data coding and decoding method comprising the following steps: (a) coding voice data sent by a channel side to get a voice information packet Ss; (b) adding packet length and timestamp information to the header of the information packet Ss to get an extended information packet Ss', and pressing the extended information packet Ss' into a voice queue; (c) performing RS error-correction coding on the information packet Ss' through an RS coding mode to get a redundant packet Rs, adding related protection information to the header of Rs to get an extended redundant packet Rs', and packaging the information packet Ss and the extended redundant packet Rs' together to generate a combined packet; and (d) at a channel receiver, recovering a lost information packet by means of RS packet-loss recovery according to the information packet and the redundant packet received. The channel receiver unpacks and calibrates the received combined packet. If the sum of the number of associated information packets SN and the number of associated redundant packets RN is greater than or equal to k, the combined packet is transferred to the packet-loss recovery operation and then decoded. According to the method, whether a wrong packet can be recovered is detected at the receiver, and a wrong packet is recovered immediately if the wrong packet can be recovered. Delay is reduced. The problem concerning voice quality caused by network packet loss, delay and jitter is repaired effectively.

Description

technical field [0001] The invention belongs to the technical field of Internet telephony, in particular to the voice processing technology of Internet telephony. Background technique [0002] With the rapid development of network technology, streaming media technology, one of the most popular Internet technologies at present, is the basis for us to share images, audio, video and other multimedia files quickly and with high fidelity. In streaming media technology, due to external factors such as the network, errors such as packet loss, out-of-order, and delayed arrival are inevitable. Voice files are the most sensitive to network quality. If there is a packet error in the audio file, such as packet loss, disorder, delay, etc., it will seriously affect the voice hearing effect of the receiving end, jitter, noise, and voice fuzzy. Can't even recognize voice content. [0003] In order to ensure that the transmitted audio can be played normally at the receiving end, it is nece...

Claims

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Application Information

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IPC IPC(8): H04L1/00
CPCH04L1/0057H04L1/0061
Inventor 李俣菡黄敬佩徐晓峰李夏宾
Owner SHENZHEN GRANDSTREAM NETWORKS TECH
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