Method, system and control server for processing audio

Abstract

The invention discloses an audio processing method, which comprises the steps as follows: a control server obtains the audio capability of a terminal through capability negotiation after the terminal is connected with the control server; and the control server transmits coded audio data to each terminal according to the audio capability. The invention further discloses an audio processing system and the control server. The audio data in the embodiment of the invention does not need audio coding decoding operation when passing the control server every time, so that the coding decoding frequency of the control server is reduced greatly; especially, the audio delay between terminals only includes network transmission, the coding at a sending terminal and the decoding at a receiving terminal when only one control server is provided, and the control server only recomposes extracting packages and group packages on the audio data, so that the delay is negligible, the interactive real-time performance between the terminals can be enhanced, audio coding decoding resource occupation by the control server can be reduced, and the cost is lowered.

Description

technical field [0001] The invention relates to the technical field of voice communication, in particular to an audio processing method, system and control server. Background technique [0002] The current conference TV products or some conference phone products mainly follow the ITU-H.323 protocol or ITU-H.320 protocol for audio processing. Among them, the equipment that realizes core audio exchange and controls multiple conference terminals is mainly MCU (Multipoint Control Unit , multipoint control unit), the MCU unit has at least MC (Multipoint Control, multipoint control) function and MP (Multipoint Process, multipoint processing) function, and can perform multi-channel mixing. For example, in a conference call, at least The telephone terminals of the three venues communicate through the MCU at the same time, and the MCU needs to mix the voices sent by each terminal into one, and then send it to the telephone terminals of each venue, so as to ensure that although the en...

AI Technical Summary

Problems solved by technology

[0011] In the process of realizing the present invention, the inventors found that the prior art has at least the following problems: each codec will increase the terminal-to-terminal audio distortion. Decoding, when the MCU mixes audio, one codec is required, resulting in two distortions; when a multipoint conference based on two MCUs is cascaded, the site terminal needs to be coded once, and two MCUs are required to be encoded twice. Decoding, resulting in three distortions; and so on, every time an MCU is added, there will be a corresponding increase in distortion

Each codec will increase the end-to-end voice delay. The reason and derivation process are consistent with the above-mentioned audio distortion, and due to the delay of audio codec, especially when at least two MCUs are cascaded, it is easy to cause lip synchronization. problem, it will take a lot of manpower and material resources to solve this problem

Furthermore, for the conference terminals that join the voice conference at the same time, the MCU needs to allocate audio codec ports for each terminal. The setting of multiple ports increases the cost of multipoint conferences, especially when there are many conference sites, the MCU needs to provide a large amount of audio. codec port

Images