CN105391523A - Voice optimization transmission method and device - Google Patents
Voice optimization transmission method and device Download PDFInfo
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- CN105391523A CN105391523A CN201510957539.0A CN201510957539A CN105391523A CN 105391523 A CN105391523 A CN 105391523A CN 201510957539 A CN201510957539 A CN 201510957539A CN 105391523 A CN105391523 A CN 105391523A
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L1/00—Arrangements for detecting or preventing errors in the information received
- H04L1/0078—Avoidance of errors by organising the transmitted data in a format specifically designed to deal with errors, e.g. location
- H04L1/0083—Formatting with frames or packets; Protocol or part of protocol for error control
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L1/00—Arrangements for detecting or preventing errors in the information received
- H04L1/0078—Avoidance of errors by organising the transmitted data in a format specifically designed to deal with errors, e.g. location
- H04L1/0091—Avoidance of errors by organising the transmitted data in a format specifically designed to deal with errors, e.g. location arrangements specific to receivers, e.g. format detection
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/80—Responding to QoS
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Abstract
The invention relates to the communication field and especially relates to a voice optimization transmission method and device. Through sacrifice of a part of transmission bandwidth, each voice data packet comprises normally-read data content of the data packet, and continuous data content sent before; by calculating the difference value of two voice data packets received successively, whether the case of data packet dropout occurs can be judged; if not, only the normally-read data content of the data packet is read; and if so, the lost voice data packet is restored in the previously-sent continuous data content of the current voice data packet. In general, the method above is not adopted for the sake of data transmission efficiency and full utilization of transmission bandwidth; and however, through sacrifice of a part of transmission bandwidth, the lost data is restored as much as possible, thereby ensuring clearness and smoothness of the speech, and ensuring communication quality. Besides, in the scheme, data does not need to be stored, and efficiency can be improved.
Description
Technical field
The present invention relates to the communications field, particularly relate to a kind of voice-optimizing transmission method and device.
Background technology
Mobile radio communications system uses Wireless Ad Hoc Networks, but radio communication is usually insecure, and in the poor situation of wireless signal, packet loss is comparatively serious, and voice communication quality is a greater impact, and voice communication usually occurs Caton phenomenon.
In Real-Time Voice Transmission, because voice require stronger real-time, not very sensitive to a small amount of loss of data, therefore need and general networking transmission diverse ways.The Transmission Control Protocol that real-time makes the inapplicable band of voice transfer confirm and retransmit, usual unserviceable udp protocol, but UDP is inevitably with relatively high packet loss, how to resist packet loss and the relevant issues how to process become the focus studied in real-time speech communicating when there is packet loss phenomenon.
Packet loss treatment technology mainly contains forward error correction (FEC), intertexture, packet loss concealment etc.
Forward error correction technique is the general designation of a class channel redundancy coding, and object is the reliability improving voice data transmission, can recover the bag lost when there is indivedual random loss.This kind of coding has and simply has complexity, and it is little that simple code takies extra bandwidth, and recovery capability is poor, as parity check etc.; More complicated code restoration ability is good, takies extra bandwidth comparatively large, as RS code etc.LDPC code has good coding efficiency simultaneously, and has parameter adjustment more flexibly, easily decoded mode, is applied at present in some fields.But FEC technology has a feature, in certain packet loss limit, data can be recovered completely, but exceed this limit, then cannot recover completely.
Interleaving technology is a kind of method reducing packet loss loss.Initial data is divided into less frame, before sending, resets the order of frame, make data in each bag from staggered speech frame.Thus when there is packet loss, loss be discrete frame data, if these frames are little, little on sense of hearing impact; And also facilitate and follow-up bag-losing hide process is done to these frame losing data of comparatively disperseing, but interleaving technology easily causes larger propagation delay time.
Bag-losing hide refers to that receiving terminal is when packet loss or frame losing occur, and fills up the data of loss by certain algorithm, reduces the loss that obliterated data brings.Mainly comprise insertion and interpolation technique, insert the data referring to and lose with fixing signal substituting, interpolation refers to the short-term correlation according to known signal and voice, the data that structure is lost.
Existing interleaving technology itself does not provide redundancy and error correction, and FEC also not supported data part recover.
Summary of the invention
Technical problem to be solved by this invention is: provide a kind of voice-optimizing transmission method and the device with packet loss restoring function.
In order to solve the problems of the technologies described above, the technical solution used in the present invention is:
A kind of voice-optimizing transmission method, comprising:
Step 1, receive speech data compressed package according to natural number number consecutively; Described speech data compressed package is made up of N number of compress speech frame, is respectively 1 the first compress speech frame read for speech data and N-1 for reducing the compress speech frame of packet loss; Described N-1 is continuous print speech data before the first compress speech frame for reducing the compress speech frame of packet loss; The value of N be greater than 1 integer;
Step 2, calculate the numbering difference of the double speech data compressed package received, if described numbering difference is greater than 1 and is less than or equal to N, then according to numbering difference successively from described N-1 for reduce packet loss compress speech frame in read the compress speech frame that is connected with the first compress speech frame; If described numbering difference is greater than N, then to read after in the speech data compressed package that receives for twice the whole compress speech frames in speech data compressed package once.
Another technical scheme that the present invention adopts is:
A kind of voice data transmission device, comprises receiver module and computing module;
Described receiver module, for receiving the speech data compressed package according to natural number number consecutively; Described speech data compressed package is made up of N number of compress speech frame, is respectively 1 the first compress speech frame read for speech data and N-1 for reducing the compress speech frame of packet loss; Described N-1 is continuous print speech data before the first compress speech frame for reducing the compress speech frame of packet loss; The value of N be greater than 1 integer;
Described computing module comprises computing unit, the first reading unit and the second reading unit;
Described computing unit, for calculating the numbering difference of the double speech data compressed package received;
Described first reading unit, if be greater than 1 for described numbering difference and be less than or equal to N, then according to numbering difference successively from described N-1 for reduce packet loss compress speech frame in read the compress speech frame that is connected with the first compress speech frame;
Described second reading unit, if be greater than N for described numbering difference, then to read after in the speech data compressed package that receives for twice the whole compress speech frames in speech data compressed package once.
Beneficial effect of the present invention is: by sacrificial section transmission bandwidth, all containing the data content that this compressed package normally reads in each speech data compressed package, the continuous print data content sent before also comprising, by calculating the numbering difference of the speech data compressed package that connection receives for twice, can judge whether data-bag lost situation occurs, if there is no packet loss, only read the data content that this compressed package normally reads, if packet loss, then from current speech data compressed package before send continuous print data content reduce lose speech data compressed package.Generally for the efficiency of transfer of data, utilize transmission bandwidth fully, to adopt above-mentioned this mode, and the technical solution adopted in the present invention is mainly in order to reduce the probability of packet loss, by sacrificial section transmission bandwidth, reduce the data of losing as much as possible, ensure that lamprophonia is smooth, ensure that the quality of communication.This programme is without the need to storing data in addition, and then can provide efficiency.
Accompanying drawing explanation
Fig. 1 is the flow chart of steps of voice-optimizing transmission method of the present invention;
Fig. 2 is the structural representation of voice-optimizing transmitting device of the present invention;
Label declaration:
1, receiver module;
2, computing module; 21, computing unit; 22, the first reading unit; 23, the second reading unit.
Embodiment
By describing technology contents of the present invention in detail, realized object and effect, accompanying drawing is coordinated to be explained below in conjunction with execution mode.
The design of most critical of the present invention is: by sacrificial section transmission bandwidth, all containing the data content that this compressed package normally reads in each speech data compressed package, the continuous print data content sent before also comprising, if there is no packet loss, only read the data content that this compressed package normally reads, if packet loss, then from current speech data compressed package before send continuous print data content reduce lose speech data compressed package.
Please refer to Fig. 1, a kind of voice-optimizing transmission method provided by the invention, comprising:
Step 1, receive speech data compressed package according to natural number number consecutively; Described speech data compressed package is made up of N number of compress speech frame, is respectively 1 the first compress speech frame read for speech data and N-1 for reducing the compress speech frame of packet loss; Described N-1 is continuous print speech data before the first compress speech frame for reducing the compress speech frame of packet loss; The value of N be greater than 1 integer;
Step 2, calculate the numbering difference of the double speech data compressed package received, if described numbering difference is greater than 1 and is less than or equal to N, then according to numbering difference successively from described N-1 for reduce packet loss compress speech frame in read the compress speech frame that is connected with the first compress speech frame; If described numbering difference is greater than N, then to read after in the speech data compressed package that receives for twice the whole compress speech frames in speech data compressed package once.
From foregoing description, beneficial effect of the present invention is: by sacrificial section transmission bandwidth, all containing the data content that this compressed package normally reads in each speech data compressed package, the continuous print data content sent before also comprising, by calculating the numbering difference of the speech data compressed package that connection receives for twice, can judge whether data-bag lost situation occurs, if there is no packet loss, only read the data content that this compressed package normally reads, if packet loss, then from current speech data compressed package before send continuous print data content reduce lose speech data compressed package.Generally for the efficiency of transfer of data, utilize transmission bandwidth fully, to adopt above-mentioned this mode, and the technical solution adopted in the present invention is mainly in order to reduce the probability of packet loss, by sacrificial section transmission bandwidth, reduce the data of losing as much as possible, ensure that lamprophonia is smooth, ensure that the quality of communication.This programme is without the need to storing data in addition, and then can provide efficiency.
Further, described step 2 also comprises: if described numbering difference equals 1, then to read after in the speech data compressed package that receives for twice the first compress speech frame in speech data compressed package once.
Seen from the above description, if described numbering difference equals 1, illustrate there is no packet loss, directly can read speech data.
Further, N number of compress speech frame of described speech data compressed package adopts natural number number consecutively.
Seen from the above description, by carrying out number consecutively to N number of compress speech frame, can carry out looking for bag according to number order after being convenient to packet loss.
Further, the value of described N is 10.
Seen from the above description, according to practice process, when the value of N is 10, while percent reduction is high, efficiency of transmission is the fastest.
Consult Fig. 2, present invention also offers a kind of voice-optimizing transmitting device, comprise receiver module 1 and computing module 2;
Described receiver module 1, for receiving the speech data compressed package according to natural number number consecutively; Described speech data compressed package is made up of N number of compress speech frame, is respectively 1 the first compress speech frame read for speech data and N-1 for reducing the compress speech frame of packet loss; Described N-1 is continuous print speech data after the first compress speech frame for reducing the compress speech frame of packet loss; The value of N be greater than 1 integer;
Described computing module 2 comprises computing unit 21, first reading unit 22 and the second reading unit 23;
Described computing unit 21, for calculating the numbering difference of the double speech data compressed package received;
Described first reading unit 22, if be greater than 1 for described numbering difference and be less than or equal to N, then according to numbering difference successively from described N-1 for reduce packet loss compress speech frame in read the compress speech frame that is connected with the first compress speech frame;
Described second reading unit 23, if be greater than N for described numbering difference, then reads the whole compress speech frames in the speech data compressed package received for twice in previous speech data compressed package.
Further, described computing module 2 also comprises third reading and gets unit;
Described third reading gets unit, if equal 1 for described numbering difference, then to read after in the speech data compressed package that receives for twice the first compress speech frame in speech data compressed package once.
Seen from the above description, if described numbering difference equals 1, illustrate there is no packet loss, directly can read speech data.
Further, N number of compress speech frame of described speech data compressed package adopts natural number number consecutively.
Seen from the above description, by carrying out number consecutively to N number of compress speech frame, can carry out looking for bag according to number order after being convenient to packet loss.
Further, the value of described N is 10.
Seen from the above description, according to practice process, when the value of N is 10, while percent reduction is high, efficiency of transmission is the fastest.
Please refer to Fig. 1, embodiments of the invention one are:
The invention provides a kind of voice-optimizing transmission method, is 3 to be described for N value.
Such as: speech data compressed package is made up of 3 compress speech frames, the compress speech frame number consecutively in first speech data compressed package is 3,2,1; The compress speech frame being wherein numbered 3 is the data that this really will send, and is numbered 2, the compress speech frame of 1 for before the data that send of continuous print two speech data compressed packages of sending; Therefore, if there is not packet drop, be numbered 2, the compress speech frame of 1 is otiose, is numbered 2, looks for bag when the compress speech frame of 1 is only used to follow-up packet loss.
Suppose: the compress speech frame number consecutively in first speech data compressed package is 3,2,1; Compress speech frame number consecutively in second speech data compressed package is 4,3,2; Compress speech frame number consecutively in 3rd speech data compressed package is 5,4,3; Ensuing speech data compressed package by that analogy.
Following examples are the situations illustrating that second speech data compressed package is lost.
Step 1, receiving terminal receive the speech data compressed package according to natural number number consecutively;
Step 2, the double speech data compressed package received of receiving terminal is respectively first speech data compressed package and the 3rd speech data compressed package, now receiving terminal calculates the numbering difference of two speech data compressed packages, numbering difference is 2, be transmitting procedure and lost 1 speech data compressed package, now from the 3rd speech data compressed package from 2 for reduce packet loss compress speech frame in read the compress speech frame that is connected with the first compress speech frame, namely read numbering compress speech frame immediately, be the compress speech frame being numbered 2, owing to only losing a bag, as long as so read a compress speech frame.Such mode just can reduce lose data content.
If described numbering difference is greater than 2, then to read after in the speech data compressed package that receives for twice the whole compress speech frames in speech data compressed package once, be the compress speech frame of numbering 2 and numbering 1 in the 3rd speech data compressed package.
Value due to N is desirable large also desirable little, when value is excessive, bandwidth availability ratio is just very low, efficiency of transmission is just very slow, and when value is too small, the data of loss can not be reduced as much as possible, cause communication quality low, but drawn by the experiment of volume, when the value of N is 10 time, realize percent reduction high while efficiency of transmission the fastest.
Embodiment two
Similar with embodiment one, for N for 10;
Compress speech frame number consecutively in first speech data compressed package is 10,9,8,7,6,5,4,3,2,1;
Compress speech frame number consecutively in second speech data compressed package is 11,10,9,8,7,6,5,4,3,2;
Compress speech frame number consecutively in 3rd speech data compressed package is 12,11,10,9,8,7,6,5,4,3;
Compress speech frame number consecutively in 4th speech data compressed package is 13,12,11,10,9,8,7,6,5,4;
Compress speech frame number consecutively in 5th speech data compressed package is 14,13,12,11,10,9,8,7,6,5;
Compress speech frame number consecutively in 6th speech data compressed package is 15,14,13,12,11,10,9,8,7,6;
Compress speech frame number consecutively in 7th speech data compressed package is 16,15,14,13,12,11,10,9,8,7;
Ensuing speech data compressed package by that analogy.
The present embodiment two is that the three-five situations that speech data compressed package is lost are described.
Step 1, receiving terminal receive the speech data compressed package according to natural number number consecutively;
Step 2, the double speech data compressed package received of receiving terminal is respectively second speech data compressed package and the 6th speech data compressed package, now receiving terminal calculates the numbering difference of two speech data compressed packages, numbering difference is 4, be transmitting procedure and lost 3 speech data compressed packages, now from the 6th speech data compressed package from 9 for reduce packet loss compress speech frame in read the compress speech frame that is connected with the first compress speech frame, namely read numbering compress speech frame immediately, be the compress speech frame being numbered 14 to start, owing to lost 3 speech data compressed packages, be numbered the compress speech frame of 13 so continue to read and be numbered the compress speech frame of 12.But according to the compress speech frame being numbered 12 start play, such mode just can reduce loss data content.
If described numbering difference is greater than 10, then to read after in the speech data compressed package that receives for twice the whole compress speech frames in speech data compressed package once, such as difference is 11, be loss 10 speech data compressed packages, the content also differing from 1 speech data compressed package now reads all data in current speech data compressed package, although cannot read, due to the continuity of language, when N value is got enough large, reduction degree can reach very high.
Value due to N is desirable large also desirable little, when value is excessive, bandwidth availability ratio is just very low, efficiency of transmission is just very slow, and when value is too small, the data of loss can not be reduced as much as possible, cause communication quality low, but drawn by the experiment of volume, when the value of N is 10 time, realize percent reduction high while efficiency of transmission the fastest.
Embodiment three
Similar with embodiment one, for N for 15;
Compress speech frame number consecutively in first speech data compressed package is 15,14,13,12,11,10,9,8,7,6,5,4,3,2,1;
Compress speech frame number consecutively in second speech data compressed package is 16,15,14,13,12,11,10,9,8,7,6,5,4,3,2;
Compress speech frame number consecutively in 3rd speech data compressed package is 17,16,15,14,13,12,11,10,9,8,7,6,5,4,3;
Compress speech frame number consecutively in 4th speech data compressed package is 18,17,16,15,14,13,12,11,10,9,8,7,6,5,4;
Compress speech frame number consecutively in 5th speech data compressed package is 19,18,17,16,15,14,13,12,11,10,9,8,7,6,5;
Ensuing speech data compressed package by that analogy;
Compress speech frame number consecutively in tenth speech data compressed package is 24,23,22,21,20,19,18,17,16,15,14,13,12,11,10;
Compress speech frame number consecutively in 17 speech data compressed package is 31,30,29,28,27,26,25,24,23,22,21,20,19,18,17;
The present embodiment three is that the five-nine situations that speech data compressed package is lost are described.
Step 1, receiving terminal receive the speech data compressed package according to natural number number consecutively;
Step 2, the double speech data compressed package received of receiving terminal is respectively the 4th speech data compressed package and the tenth speech data compressed package, now receiving terminal calculates the numbering difference of two speech data compressed packages, numbering difference is 6, be transmitting procedure and lost 5 speech data compressed packages, now from the tenth speech data compressed package from 14 for reduce packet loss compress speech frame in read the compress speech frame that is connected with the first compress speech frame, namely read numbering compress speech frame immediately, be the compress speech frame being numbered 23 to start, owing to lost 5 speech data compressed packages, so continue reading to be numbered 22, 21, 20, the compress speech frame of 19.Then according to the compress speech frame being numbered 19 start play, such mode just can reduce loss data content.
If described numbering difference is greater than 15, then to read after in the speech data compressed package that receives for twice the whole compress speech frames in speech data compressed package once, the speech data compressed package received for such as twice is respectively first speech data compressed package and the 17 speech data compressed package, this time difference value is 16, be loss 15 speech data compressed packages, now read all data in current speech data compressed package, be and be numbered 30, 29, 28, 27, 26, 25, 24, 23, 22, 21, 20, 19, 18, the compress speech frame of 17, although the content also differing from 1 speech data compressed package cannot read, but due to the continuity of language, when N value is got enough large, reduction degree can reach very high.
Value due to N is desirable large also desirable little, when value is excessive, bandwidth availability ratio is just very low, efficiency of transmission is just very slow, and when value is too small, the data of loss can not be reduced as much as possible, cause communication quality low, but drawn by the experiment of volume, when the value of N is 10 time, realize percent reduction high while efficiency of transmission the fastest.
In sum, a kind of voice-optimizing transmission method provided by the invention and device, by sacrificial section transmission bandwidth, all containing the data content that this compressed package normally reads in each speech data compressed package, the continuous print data content sent before also comprising, by calculating the numbering difference of the speech data compressed package that connection receives for twice, can judge whether data-bag lost situation occurs, if there is no packet loss, only read the data content that this compressed package normally reads, if packet loss, then from current speech data compressed package before send continuous print data content reduce lose speech data compressed package.Generally for the efficiency of transfer of data, utilize transmission bandwidth fully, to adopt above-mentioned this mode, and the technical solution adopted in the present invention is mainly in order to reduce the probability of packet loss, by sacrificial section transmission bandwidth, reduce the data of losing as much as possible, ensure that lamprophonia is smooth, ensure that the quality of communication.This programme is without the need to storing data in addition, and then can provide efficiency.
The foregoing is only embodiments of the invention; not thereby the scope of the claims of the present invention is limited; every equivalents utilizing specification of the present invention and accompanying drawing content to do, or be directly or indirectly used in relevant technical field, be all in like manner included in scope of patent protection of the present invention.
Claims (8)
1. a voice-optimizing transmission method, is characterized in that, comprising:
Step 1, receive speech data compressed package according to natural number number consecutively; Described speech data compressed package is made up of N number of compress speech frame, is respectively 1 the first compress speech frame read for speech data and N-1 for reducing the compress speech frame of packet loss; Described N-1 is continuous print speech data before the first compress speech frame for reducing the compress speech frame of packet loss; The value of N be greater than 1 integer;
Step 2, calculate the numbering difference of the double speech data compressed package received, if described numbering difference is greater than 1 and is less than or equal to N, then according to numbering difference successively from described N-1 for reduce packet loss compress speech frame in read the compress speech frame that is connected with the first compress speech frame; If described numbering difference is greater than N, then to read after in the speech data compressed package that receives for twice the whole compress speech frames in speech data compressed package once.
2. voice-optimizing transmission method according to claim 1, it is characterized in that, described step 2 also comprises: if described numbering difference equals 1, then to read after in the speech data compressed package that receives for twice the first compress speech frame in speech data compressed package once.
3. voice-optimizing transmission method according to claim 1, is characterized in that, N number of compress speech frame of described speech data compressed package adopts natural number number consecutively.
4. voice-optimizing transmission method according to claim 1, is characterized in that, the value of described N is 10.
5. a voice-optimizing transmitting device, is characterized in that, comprises receiver module and computing module;
Described receiver module, for receiving the speech data compressed package according to natural number number consecutively; Described speech data compressed package is made up of N number of compress speech frame, is respectively 1 the first compress speech frame read for speech data and N-1 for reducing the compress speech frame of packet loss; Described N-1 is continuous print speech data before the first compress speech frame for reducing the compress speech frame of packet loss; The value of N be greater than 1 integer;
Described computing module comprises computing unit, the first reading unit and the second reading unit;
Described computing unit, for calculating the numbering difference of the double speech data compressed package received;
Described first reading unit, if be greater than 1 for described numbering difference and be less than or equal to N, then according to numbering difference successively from described N-1 for reduce packet loss compress speech frame in read the compress speech frame that is connected with the first compress speech frame;
Described second reading unit, if be greater than N for described numbering difference, then to read after in the speech data compressed package that receives for twice the whole compress speech frames in speech data compressed package once.
6. voice-optimizing transmitting device according to claim 5, is characterized in that, described computing module also comprises third reading and gets unit;
Described third reading gets unit, if equal 1 for described numbering difference, then to read after in the speech data compressed package that receives for twice the first compress speech frame in speech data compressed package once.
7. voice-optimizing transmitting device according to claim 5, is characterized in that, N number of compress speech frame of described speech data compressed package adopts natural number number consecutively.
8. voice-optimizing transmitting device according to claim 5, is characterized in that, the value of described N is 10.
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| CN106530745B (en) * | 2016-12-13 | 2019-08-20 | 青岛海信网络科技股份有限公司 | A kind of data packet transmission of earth magnetism vehicle detection and processing method and main controller |
| CN110557226A (en) * | 2019-09-05 | 2019-12-10 | 北京云中融信网络科技有限公司 | Audio transmission method and device |
| CN112908346A (en) * | 2019-11-19 | 2021-06-04 | 中国移动通信集团山东有限公司 | Packet loss recovery method and device, electronic equipment and computer readable storage medium |
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