CN103248774A - VoIP server synchronous sound mixing method and system - Google Patents

VoIP server synchronous sound mixing method and system Download PDF

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Publication number
CN103248774A
CN103248774A CN2012100316310A CN201210031631A CN103248774A CN 103248774 A CN103248774 A CN 103248774A CN 2012100316310 A CN2012100316310 A CN 2012100316310A CN 201210031631 A CN201210031631 A CN 201210031631A CN 103248774 A CN103248774 A CN 103248774A
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data
voip
audio mixing
packet
voip client
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CN103248774B (en
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张海东
陈剑勇
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Abstract

The invention is applicable to the technical field of computers, and provides a VoIP (Voice Over Internet Protocol) server synchronous sound mixing method and system. The method comprises the following steps: detecting jiggle cache areas for distributing all VoIP client sides at advance; when data of all the VoIP client sides exits in the corresponding jiggle cache areas, acquiring data packages form the jiggle cache areas corresponding to all the VoIP client sides; decoding the data packages acquired, and carrying out the sound mixing processing on the decoded data according to the preset sound mixing algorithm; and sending the sound-mixed data to all the VoIP client sides. Through the arrangement of the jiggle cache areas, the data jiggle during the sound mixing process is eliminated effectively, and the sound mixing effect is improved, which increases the conversation experiment of VoIP users.

Description

A kind of VoIP server sync sound mixing method and system
Technical field
The invention belongs to field of computer technology, relate in particular to a kind of VoIP server sync sound mixing method and system.
Background technology
Along with the progress of network technology, (Voice over Internet Protocol is VoIP) because its cheap cost of the phone call and good network amalgamation have obtained application more and more widely to the networking telephone.TeleConference Bridge based on internet (IP network) can utilize the original network line of enterprise and equipment to build, use also very convenient, thereby reduced entreprise cost.
Sound mixing is the important component part of voip phone conference system and multimedia conference system, and present audio mixing scheme has adopted traditional decoding-audio mixing-coding mode.Yet, control requires than higher because speech conference system is to time delay, most of voip phone conference systems all are to adopt User Datagram Protocol (UserDatagram Protocol, UDP) send packets of audio data, because order was identical when the transmission mechanism of UDP can not guarantee order that packet arrives and send, and can not guarantee also that in transmission course packet one arrives the destination surely, thereby cause voip phone conference system voice data to be shaken, the audio mixing effect is second-rate, at present, there is not a kind of effective method to solve this problem as yet.
Summary of the invention
The purpose of the embodiment of the invention is to provide a kind of VoIP server sync sound mixing method and system, is intended to solve because prior art can't provide a kind of effective VoIP server sync sound mixing method, causes the problem of VoIP server audio mixing weak effect.
The embodiment of the invention is achieved in that a kind of VoIP server sync sound mixing method, and described method comprises the steps:
Detecting is the dithering cache district that each VoIP client is distributed in advance;
When all having data in the dithering cache district that detects all VoIP client correspondences, from the dithering cache district of each VoIP client correspondence, obtain a packet separately;
All packets that obtain are decoded, according to default audio mixing algorithm decoded data are carried out audio mixing and handle;
Data behind the audio mixing are sent to all VoIP clients.
Another purpose of the embodiment of the invention is to provide a kind of VoIP server sync mixer system, it is characterized in that described system comprises:
The buffer area detecting unit is for detection of being the dithering cache district that each VoIP client is distributed in advance;
The first packet acquiring unit is used for obtaining a packet separately from the dithering cache district of each VoIP client correspondence when all there are data in the dithering cache district that detects all VoIP client correspondences;
The audio mixing processing unit is used for all packets that obtain are decoded, and according to default audio mixing algorithm decoded data is carried out audio mixing and handles; And
Data transmission unit is used for the data behind the audio mixing are sent to all VoIP clients.
It is the dithering cache district that each VoIP client is distributed in advance that the embodiment of the invention detects, when all having data in the dithering cache district that detects all VoIP client correspondences, from the dithering cache district of each VoIP client correspondence, obtain a packet separately, all packets that obtain are decoded, according to default audio mixing algorithm decoded data being carried out audio mixing handles, data behind the audio mixing are sent to all VoIP clients, realized the audio mixing of VoIP speech data, thereby by being set, the dithering cache district effectively eliminated the shake of data in the audio mixing process, improved the audio mixing quality, thereby the conversation that has improved the voip user is experienced.
Description of drawings
Fig. 1 is the realization flow figure of the VoIP server sync sound mixing method that provides of the embodiment of the invention one;
Fig. 2 is the realization flow figure of the VoIP server sync sound mixing method that provides of the embodiment of the invention two;
Fig. 3 is the structure chart of the VoIP server sync mixer system that provides of the embodiment of the invention three;
Fig. 4 is the structure chart of the VoIP server sync mixer system that provides of the embodiment of the invention four.
Embodiment
In order to make purpose of the present invention, technical scheme and advantage clearer, below in conjunction with drawings and Examples, the present invention is further elaborated.Should be appreciated that specific embodiment described herein only in order to explaining the present invention, and be not used in restriction the present invention.
Below in conjunction with specific embodiment specific implementation of the present invention is described in detail:
Embodiment one:
Fig. 1 shows the realization flow of the VoIP server sync sound mixing method that the embodiment of the invention one provides, and details are as follows:
In step S101, detecting is the dithering cache district that each VoIP client is distributed in advance.
In embodiments of the present invention, the speech data that sends over for buffer memory VoIP client, should be that each VoIP client is distributed a corresponding cache district in advance, to be used for the buffer memory speech data, speech data is sorted, use udp protocol to send the caused data dithering problem of data thereby effectively eliminate.
In concrete implementation process, the dithering cache district can represent with the form of chained list, thereby makes things convenient for storage and the deletion of data, improves the utilance of buffer area.Particularly, each node is a structure, and the information that this structure is preserved comprises: the pointer of the duration of the length of data, timestamp, data correspondence (notebook data comprises data how long, is generally 20ms), sensing storage real data.After receiving packet, can put it in the corresponding dithering cache district according to the VoIP client identification information of its head.When putting into the dithering cache district, if the dithering cache district is full, to be the oldest then, the packet of (the oldest timestamp minimum that namely refers to of what is called) be lost, size according to timestamp is inserted this packet in its suitable position, all is by from small to large tactic of timestamp to guarantee packets all in the dithering cache district.
In step S102, when all having data in the dithering cache district that detects all VoIP client correspondences, from the dithering cache district of each VoIP client correspondence, obtain a packet separately.
In embodiments of the present invention, detect when all having data in the dithering cache district of all VoIP client correspondences, from the dithering cache district of each VoIP client correspondence, obtain a packet separately, handle to be used for follow-up sound mixing.
In step S103, all packets that obtain are decoded, according to default audio mixing algorithm decoded data are carried out audio mixing and handle.
In embodiments of the present invention, because speech data on being sent to the VoIP server time, encodes, therefore, and should be relatively to its decoding, the decoding back is carried out audio mixing according to default audio mixing algorithm to decoded data and is handled.Particularly, default audio mixing algorithm can be average audio mixing algorithm, clamp audio mixing algorithm etc., at this not in order to limit the present invention.
In step S104, the data behind the audio mixing are sent to all VoIP clients.
The embodiment of the invention is by being each VoIP client distribution dithering cache district in advance, the packet that each VoIP client of buffer memory sends, all packets that obtain are decoded, according to default audio mixing algorithm decoded data being carried out audio mixing handles, realized the audio mixing of VoIP speech data, thereby effectively eliminated the shake of data in the audio mixing process, improved the effect of audio mixing, thereby the conversation that has improved the voip user is experienced.
Embodiment two:
Fig. 2 shows the realization flow of the VoIP server sync sound mixing method that the embodiment of the invention two provides, and details are as follows:
In step S201, detecting is the dithering cache district that each VoIP client is distributed in advance.
In embodiments of the present invention, the speech data that sends over for buffer memory VoIP client, should be that each VoIP client is distributed a corresponding cache district in advance, to be used for the buffer memory speech data, speech data is sorted, use udp protocol to send the caused data dithering problem of data thereby effectively eliminate.
In step S202, judge whether to surpass default detection time this detection time, be execution in step S207 then, otherwise execution in step S203.
In embodiments of the present invention, to carrying out timing detection time, judge whether surpass default detection time each detection time, can produce very big influence to the audio mixing effect detection time.If detection time is oversize, because the phenomenon of packet loss can appear in certain road (data that send from certain VoIP client) sometimes, time-delay will become greatly, as if detection time weak point will be because of the speech data of each VoIP client hash and cause the audio mixing deleterious too.Draw through a large amount of experiment tests, be preferably the 500-1000 millisecond default detection time.
In step S203, judge whether to detect in the dithering cache district of all VoIP client correspondences and all have data, be execution in step S204 then, otherwise execution in step S201.
In step S204, from the dithering cache district of each VoIP client correspondence, obtain a packet separately.
In step S205, all packets that obtain are decoded, according to default audio mixing algorithm decoded data are carried out audio mixing and handle.
In step S206, the data behind the audio mixing are sent to all VoIP clients.
In embodiments of the present invention, its concrete enforcement of step S204 to S206 is identical with step S102 to S104 in the enforcement one, does not give unnecessary details at this.
In step S207, from the dithering cache district of each VoIP client correspondence of having packet, obtain packet.
In embodiments of the present invention, surpass default detection time when the cycle detection time, do not detect when all having data in the dithering cache district of all VoIP client correspondences, from the dithering cache district of each VoIP client correspondence of having packet, obtain packet.In the concrete implementation process of implementing, also can be from the dithering cache district of all VoIP client correspondences, to obtain packet, the result who just never exists the buffer area of data to obtain is sky.Trigger step S205 after obtaining packet, all packets that obtain are decoded, according to default audio mixing algorithm decoded data are carried out audio mixing and handle.
The embodiment of the invention is by being each VoIP client distribution dithering cache district in advance, the data that receive are adjusted, on the other hand by rationally be set detection time, prevent losing and speech data hash too of speech data, effectively eliminate the shake of data in the audio mixing process, improved the audio mixing quality.
One of ordinary skill in the art will appreciate that all or part of step that realizes in above-described embodiment method is to instruct relevant hardware to finish by program, described program can be stored in the computer read/write memory medium, described storage medium is as ROM/RAM, disk, CD etc.
Embodiment three:
Fig. 3 shows the structure of the VoIP server sync mixer system that the embodiment of the invention three provides, and for convenience of explanation, only shows the part relevant with the embodiment of the invention, comprising:
It is the dithering cache district that each VoIP client is distributed in advance that buffer area detecting unit 31 detects.
In embodiments of the present invention, the speech data that sends over for buffer memory VoIP client, should be that each VoIP client is distributed a corresponding cache district in advance, to be used for the buffer memory speech data, speech data is sorted, use udp protocol to send the caused data dithering problem of data thereby effectively eliminate.
The first packet acquiring unit 32 obtains a packet separately from the dithering cache district of each VoIP client correspondence when all having data in the dithering cache district that detects all VoIP client correspondences.
In embodiments of the present invention, detect when all having data in the dithering cache district of all VoIP client correspondences, from the dithering cache district of each VoIP client correspondence, obtain a packet separately, handle to be used for follow-up audio mixing.
33 pairs of all packets that obtain of audio mixing processing unit are decoded, and according to default audio mixing algorithm decoded data are carried out audio mixing and handle.
In embodiments of the present invention, because speech data on being sent to the VoIP server time, encodes, therefore, and should be relatively to its decoding, the decoding back is carried out audio mixing according to default audio mixing algorithm to decoded data and is handled.Particularly, default audio mixing algorithm can be average audio mixing algorithm, clamp audio mixing algorithm etc., at this not in order to limit the present invention.
Data transmission unit 34 sends to all VoIP clients with the data behind the audio mixing.
In the invention process, the embodiment of each unit is corresponding with each step among the embodiment one, does not repeat them here.
The embodiment of the invention is by being each VoIP client distribution dithering cache district in advance, the packet that each VoIP client of buffer memory sends, all packets that obtain are decoded, according to default audio mixing algorithm decoded data being carried out audio mixing handles, realized the audio mixing of VoIP speech data, thereby effectively eliminated the shake of data in the audio mixing process, improved the effect of audio mixing, thereby the conversation that has improved the voip user is experienced.
Embodiment four:
Fig. 4 shows the structure of the VoIP server sync mixer system that the embodiment of the invention four provides, and for convenience of explanation, only shows the part relevant with the embodiment of the invention, comprising:
It is the dithering cache district that each VoIP client is distributed in advance that buffer area detecting unit 41 detects.
Detection time, judging unit 42 judged whether surpass default detection time this detection time.
In embodiments of the present invention, when detection time judging unit 42 judge and surpass default detection time this detection time, trigger the second packet acquiring unit 46 and carry out the step of from the dithering cache district of each VoIP client correspondence of having packet, obtaining packet.
The first packet acquiring unit 43 obtains a packet separately from the dithering cache district of each VoIP client correspondence when all having data in the dithering cache district that detects all VoIP client correspondences.
In embodiments of the present invention, judging unit 42 judgements do not have above default detection time this detection time when detection time, and detect when all having data in the dithering cache district of all VoIP client correspondences, the first packet acquiring unit 43 obtains a packet separately from the dithering cache district of each VoIP client correspondence.
44 pairs of all packets that obtain of audio mixing processing unit are decoded, and according to default audio mixing algorithm decoded data are carried out audio mixing and handle.
Data transmission unit 45 sends to all VoIP clients with the data behind the audio mixing.
The second packet acquiring unit 46 obtained packet from the dithering cache district of each VoIP client correspondence of having packet when described detection time, judging unit 42 judged that surpass default detection time this detection time.
In embodiments of the present invention, surpass default detection time when the cycle detection time, do not detect when all having data in the dithering cache district of all VoIP client correspondences, from the dithering cache district of each VoIP client correspondence of having packet, obtain packet.In the concrete implementation process of implementing, also can be from the dithering cache district of all VoIP client correspondences, to obtain packet, the result who just never exists the buffer area of data to obtain is sky.Therefore, the second packet acquiring unit 46 and the first packet acquiring unit 43 can be finished by a unit in specific implementation process, at this not in order to limit the present invention.
In the invention process, the embodiment of each unit is corresponding with each step among the embodiment two, does not repeat them here.
It is the dithering cache district that each VoIP client is distributed in advance that the embodiment of the invention detects, when all having data in the dithering cache district that detects all VoIP client correspondences, from the dithering cache district of each VoIP client correspondence, obtain a packet separately, all packets that obtain are decoded, according to default audio mixing algorithm decoded data being carried out audio mixing handles, data behind the audio mixing are sent to all VoIP clients, thereby by being each VoIP client distribution dithering cache district in advance, the data that receive are adjusted, on the other hand by rationally be set detection time, prevent losing and speech data hash too of speech data, effectively eliminate the shake of data in the audio mixing process, improved the audio mixing quality.
The above only is preferred embodiment of the present invention, not in order to limiting the present invention, all any modifications of doing within the spirit and principles in the present invention, is equal to and replaces and improvement etc., all should be included within protection scope of the present invention.

Claims (10)

1. a VoIP server sync sound mixing method is characterized in that described method comprises the steps:
Detecting is the dithering cache district that each VoIP client is distributed in advance;
When all having data in the dithering cache district that detects all VoIP client correspondences, from the dithering cache district of each VoIP client correspondence, obtain a packet separately;
All packets that obtain are decoded, according to default audio mixing algorithm decoded data are carried out audio mixing and handle;
Data behind the audio mixing are sent to all VoIP clients.
2. the method for claim 1 is characterized in that, described packet comprises the duration of length, timestamp and the data correspondence of data in the packet.
3. the method for claim 1, it is characterized in that, describedly obtain separately from the dithering cache district of each VoIP client correspondence before the step of a packet when all having data in the dithering cache district that detects all VoIP client correspondences, described method also comprises:
Judge and whether surpass default detection time this detection time;
Do not have to surpass default detection time when this detection time, and detect when all having data in the dithering cache district of all VoIP client correspondences, carry out the described step of from the dithering cache district of each VoIP client correspondence, obtaining a packet separately.
4. method as claimed in claim 3 is characterized in that, described method also comprises the steps:
When surpass default detection time this detection time, from the dithering cache district of each VoIP client correspondence of having packet, obtain packet, and jump to and described all packets that obtain are decoded, according to default audio mixing algorithm decoded data are carried out the step that audio mixing is handled.
5. as claim 3 or 4 described methods, it is characterized in that be the 500-1000 millisecond described default detection time.
6. the method for claim 1 is characterized in that, described audio mixing algorithm is clamp audio mixing algorithm.
7. VoIP server sync mixer system is characterized in that described system comprises:
The buffer area detecting unit is for detection of being the dithering cache district that each VoIP client is distributed in advance;
The first packet acquiring unit is used for obtaining a packet separately from the dithering cache district of each VoIP client correspondence when all there are data in the dithering cache district that detects all VoIP client correspondences;
The audio mixing processing unit is used for all packets that obtain are decoded, and according to default audio mixing algorithm decoded data is carried out audio mixing and handles; And
Data transmission unit is used for the data behind the audio mixing are sent to all VoIP clients.
8. system as claimed in claim 7 is characterized in that, described system also comprises:
Detection time, judging unit was used for judging whether surpass default detection time this detection time.
9. system as claimed in claim 8 is characterized in that, described system also comprises:
The second packet acquiring unit is used for when described detection time of judgment unit judges when this time surpass the detection time of presetting detection time, obtains packet from the dithering cache district of each VoIP client correspondence of having packet.
10. system as claimed in claim 8 or 9 is characterized in that be the 500-1000 millisecond described default detection time.
CN201210031631.0A 2012-02-13 2012-02-13 VoIP server synchronous sound mixing method and system Expired - Fee Related CN103248774B (en)

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Cited By (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN104409079A (en) * 2014-11-03 2015-03-11 北京有恒斯康通信技术有限公司 Method and device for audio superposition
CN103701624B (en) * 2013-12-31 2017-06-06 广东公信智能会议股份有限公司 A kind of voice data sound mixing method and device
CN107195308A (en) * 2017-04-14 2017-09-22 苏州科达科技股份有限公司 Sound mixing method, the apparatus and system of audio/video conference system
CN107800902A (en) * 2017-09-15 2018-03-13 北京容联易通信息技术有限公司 The sound mixing method and system of multi-path voice
CN114629998A (en) * 2022-03-04 2022-06-14 太仓市同维电子有限公司 Device and method for improving conference bridge audio mixing quality

Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN101282386A (en) * 2008-05-22 2008-10-08 中山大学 Method for forwarding synchronous mixed audio of VOIP server terminal
CN101577609A (en) * 2008-05-09 2009-11-11 深圳富泰宏精密工业有限公司 Method and device for processing Internet voice protocol packet

Patent Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN101577609A (en) * 2008-05-09 2009-11-11 深圳富泰宏精密工业有限公司 Method and device for processing Internet voice protocol packet
CN101282386A (en) * 2008-05-22 2008-10-08 中山大学 Method for forwarding synchronous mixed audio of VOIP server terminal

Cited By (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN103701624B (en) * 2013-12-31 2017-06-06 广东公信智能会议股份有限公司 A kind of voice data sound mixing method and device
CN104409079A (en) * 2014-11-03 2015-03-11 北京有恒斯康通信技术有限公司 Method and device for audio superposition
CN107195308A (en) * 2017-04-14 2017-09-22 苏州科达科技股份有限公司 Sound mixing method, the apparatus and system of audio/video conference system
CN107195308B (en) * 2017-04-14 2021-03-16 苏州科达科技股份有限公司 Audio mixing method, device and system of audio and video conference system
CN107800902A (en) * 2017-09-15 2018-03-13 北京容联易通信息技术有限公司 The sound mixing method and system of multi-path voice
CN114629998A (en) * 2022-03-04 2022-06-14 太仓市同维电子有限公司 Device and method for improving conference bridge audio mixing quality

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