CN105391523B - A kind of voice-optimizing transmission method and device - Google Patents
A kind of voice-optimizing transmission method and device Download PDFInfo
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- CN105391523B CN105391523B CN201510957539.0A CN201510957539A CN105391523B CN 105391523 B CN105391523 B CN 105391523B CN 201510957539 A CN201510957539 A CN 201510957539A CN 105391523 B CN105391523 B CN 105391523B
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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 present invention relates to the communications field more particularly to a kind of voice-optimizing transmission methods and device.By sacrificing fractional transmission bandwidth, the data content normally read containing the compressed package in each voice data compressed package, further include the continuous data content sent before, the number difference of voice data compressed package received twice by calculating connection, it can decide whether that data-bag lost situation occurs, if without packet loss, only read the data content that the compressed package is normally read, if packet loss, the voice data compressed package of loss is restored in the continuous data content sent before current speech data compressed package.It is that will not use above-mentioned this mode adequately using transmission bandwidth generally for the efficiency of data transmission, and the present invention restores the data of loss, ensures that lamprophonia is smooth, ensure that the quality of communication as far as possible by sacrificing fractional transmission bandwidth.In addition this programme is without storing data, and then can provide efficiency.
Description
Technical field
The present invention relates to the communications field more particularly to a kind of voice-optimizing transmission methods and device.
Background technology
Mobile radio communications system uses Wireless Ad Hoc Networks, but wireless communication is often insecure, wireless
In the case of dtr signal, packet loss is more serious, and voice communication quality is a greater impact, and voice communication interim card usually occurs and shows
As.
In Real-Time Voice Transmission, since voice requires stronger real-time, be not to a small amount of loss of data it is very sensitive,
Therefore the method different with general networking transmission is needed.Real-time causes voice transfer not apply to the TCP associations for confirming and retransmitting
View, usually using insecure udp protocol, but UDP is inevitably with relatively high packet loss, how to resist packet loss and
The relevant issues how handled when packet loss phenomenon occurs become the hot spot studied in real-time speech communicating.
Packet loss treatment technology mainly has forward error correction (FEC), intertexture, packet loss concealment etc..
Forward error correction technique is the general designation of a kind of channel redundancy coding, it is therefore intended that improves the reliable of voice data transmission
Property, the packet lost can be restored when indivedual random loss occur.This kind of coding has and simply has complexity, and simple code occupies extra band
Wide small, recovery capability is poor, such as even-odd check；More complicated code restoration ability is good, and occupancy extra bandwidth is larger, such as RS codes
Deng.LDPC code has preferable coding efficiency simultaneously, and has more flexible parameter adjustment, convenient decoded mode, at present one
A little fields are promoted and applied.But all there are one features for FEC technologies, and in certain packet loss limit, data can be restored completely, but
More than the limit, then can not restore completely.
Interleaving technology is a kind of method for reducing packet loss loss.Initial data is divided into smaller frame, before sending, is reset
The sequence of frame makes the data in each packet from speech frame staggeredly.So as to which when packet loss occurs, loss is discrete frame
Data if these frames are seldom, influence the sense of hearing little；And it is also convenient for doing subsequent lose to the frame losing data that these relatively disperse
Packet hides processing, but interleaving technology easily causes larger propagation delay time.
Bag-losing hide refers to that when having occurred and that packet loss or frame losing, the number of loss is filled up by certain algorithm for receiving terminal
According to the loss of data band is lost in reduction.Main to include insertion and interpolation technique, insertion refers to be lost with fixed signal substituting
Data, interpolation refers to the short-term correlation according to known signal and voice, constructs the data of loss.
Existing interleaving technology does not provide redundancy and error correction in itself, and FEC does not support the part of data to restore yet.
Invention content
The technical problems to be solved by the invention are：A kind of voice-optimizing transmission method with packet loss restoring function is provided
And device.
In order to solve the above-mentioned technical problem, the technical solution adopted by the present invention is：
A kind of voice-optimizing transmission method, including：
Step 1 is received according to the voice data compressed package of natural number number consecutively；The voice data compressed package is by N number of
Compress speech frame forms, and respectively 1 is used to restore packet loss for the first compress speech frame of voice data reading and N-1
Compress speech frame；The N-1 are continuous voice data before the first compress speech frame for restoring the compress speech frame of packet loss；
The value of N is the integer more than 1；
The number difference of voice data compressed package that step 2, calculating receive twice in succession, if the number difference is more than
1 and less than or equal to N, then it is used for successively from the N-1 according to number difference in the compress speech frame for restoring packet loss and the first language
The compress speech frame of sound condensed frame connection starts to read；If the number difference is more than N, the voice number received twice is read
According to whole compress speech frames in primary voice data compressed package rear in compressed package.
Another technical solution that the present invention uses for：
A kind of voice data transmission device, including receiving module and computing module；
The receiving module, for receiving the voice data compressed package according to natural number number consecutively；The voice data
Compressed package is made of N number of compress speech frame, respectively 1 the first compress speech frame and N-1 use for voice data reading
In the compress speech frame of reduction packet loss；The N-1 compress speech frames for being used to restore packet loss is connect before the first compress speech frame
Continuous voice data；The value of N is the integer more than 1；
The computing module includes computing unit, the first reading unit and the second reading unit；
The computing unit, for calculating the number difference of voice data compressed package received twice in succession；
First reading unit, if being more than 1 and less than or equal to N for the number difference, according to number difference according to
It is secondary to be read since the N-1 are used for the compress speech frame being connect in the compress speech frame for restoring packet loss with the first compress speech frame
It takes；
Second reading unit if being more than N for the number difference, reads the voice data pressure received twice
Whole compress speech frames after in contracting packet in primary voice data compressed package.
The beneficial effects of the present invention are：By sacrificing fractional transmission bandwidth, contain in each voice data compressed package
The data content that the compressed package is normally read further includes the continuous data content sent before, is connect twice by calculating connection
The number difference of the voice data compressed package received, it can be determined that data-bag lost situation whether occurs, if without packet loss, only
The data content that the compressed package is normally read is read, if packet loss, what is sent before current speech data compressed package is continuous
Data content in restore the voice data compressed package of loss.Generally for the efficiency of data transmission, conveyor is adequately utilized
Width is will not to use above-mentioned this mode, and the technical solution adopted in the present invention is primarily to the probability of reduction packet loss, leads to
Sacrifice fractional transmission bandwidth is crossed, restores the data of loss as far as possible, ensures that lamprophonia is smooth, ensure that the quality of communication.
In addition this programme is without storing data, and then can provide efficiency.
Description of the drawings
Fig. 1 is the step flow chart of the voice-optimizing transmission method of the present invention；
Fig. 2 is the structure diagram of the voice-optimizing transmitting device of the present invention；
Label declaration：
1st, receiving module；
2nd, computing module；21st, computing unit；22nd, the first reading unit；23rd, the second reading unit.
Specific embodiment
For the technology contents that the present invention will be described in detail, the objects and the effects, below in conjunction with embodiment and coordinate attached
Figure is explained.
The design of most critical of the present invention is：By sacrificing fractional transmission bandwidth, contain in each voice data compressed package
The data content for having the compressed package normally to read further includes the continuous data content sent before, if without packet loss, only reads
The data content that the compressed package is normally read, if packet loss, the continuous number of transmission before current speech data compressed package
According to the voice data compressed package that loss is restored in content.
Please refer to Fig. 1, a kind of voice-optimizing transmission method provided by the invention, including：
Step 1 is received according to the voice data compressed package of natural number number consecutively；The voice data compressed package is by N number of
Compress speech frame forms, and respectively 1 is used to restore packet loss for the first compress speech frame of voice data reading and N-1
Compress speech frame；The N-1 are continuous voice data before the first compress speech frame for restoring the compress speech frame of packet loss；
The value of N is the integer more than 1；
The number difference of voice data compressed package that step 2, calculating receive twice in succession, if the number difference is more than
1 and less than or equal to N, then it is used for successively from the N-1 according to number difference in the compress speech frame for restoring packet loss and the first language
The compress speech frame of sound condensed frame connection starts to read；If the number difference is more than N, the voice number received twice is read
According to whole compress speech frames in primary voice data compressed package rear in compressed package.
As can be seen from the above description, the beneficial effects of the present invention are：By sacrificing fractional transmission bandwidth, each voice data
The data content normally read containing the compressed package in compressed package further includes the continuous data content sent before, passes through
Calculate the number difference of voice data compressed package that connection receives twice, it can be determined that data-bag lost situation whether occurs,
If without packet loss, the data content that the compressed package is normally read only is read, if packet loss, from current speech data compressed package
The voice data compressed package of loss is restored in the continuous data content sent before.Generally for the efficiency of data transmission, fill
That divides utilizes transmission bandwidth, is that will not use above-mentioned this mode, and the technical solution adopted in the present invention is primarily to drop
The probability of low packet loss by sacrificing fractional transmission bandwidth, restores the data of loss, ensures that lamprophonia is smooth, protect as far as possible
The quality of communication is demonstrate,proved.In addition this programme is without storing data, and then can provide efficiency.
Further, the step 2 further includes：If the number difference is equal to 1, the voice number received twice is read
According to the first compress speech frame in primary voice data compressed package rear in compressed package.
Seen from the above description, if the number difference is equal to 1, illustrate no packet loss, voice data can be directly read.
Further, N number of compress speech frame of the voice data compressed package uses natural number number consecutively.
Seen from the above description, by N number of compress speech frame carry out number consecutively, convenient for after packet loss can according to number it is suitable
Sequence carries out looking for packet.
Further, the value of the N is 10.
Seen from the above description, according to practice process, when the value of N is 10, efficiency of transmission while reduction rate is high
It is most fast.
Referring to Fig.2, the present invention also provides a kind of voice-optimizing transmitting device, including receiving module 1 and computing module 2；
The receiving module 1, for receiving the voice data compressed package according to natural number number consecutively；The voice data
Compressed package is made of N number of compress speech frame, respectively 1 the first compress speech frame and N-1 use for voice data reading
In the compress speech frame of reduction packet loss；The N-1 compress speech frames for being used to restore packet loss is connect after the first compress speech frame
Continuous voice data；The value of N is the integer more than 1；
The computing module 2 includes computing unit 21, the first reading unit 22 and the second reading unit 23；
The computing unit 21, for calculating the number difference of voice data compressed package received twice in succession；
First reading unit 22, if being more than 1 and less than or equal to N for the number difference, according to number difference
Successively since the N-1 are used for the compress speech frame being connect in the compress speech frame for restoring packet loss with the first compress speech frame
It reads；
Second reading unit 23 if being more than N for the number difference, reads the voice data received twice
Whole compress speech frames in compressed package in previous voice data compressed package.
Further, the computing module 2 further includes third reading unit；
The third reading unit if being equal to 1 for the number difference, reads the voice data pressure received twice
The first compress speech frame after in contracting packet in primary voice data compressed package.
Seen from the above description, if the number difference is equal to 1, illustrate no packet loss, voice data can be directly read.
Further, N number of compress speech frame of the voice data compressed package uses natural number number consecutively.
Seen from the above description, by N number of compress speech frame carry out number consecutively, convenient for after packet loss can according to number it is suitable
Sequence carries out looking for packet.
Further, the value of the N is 10.
Seen from the above description, according to practice process, when the value of N is 10, efficiency of transmission while reduction rate is high
It is most fast.
Fig. 1 is please referred to, the embodiment of the present invention one is：
The present invention provides a kind of voice-optimizing transmission method, is illustrated so that N values are 3 as an example.
Such as：Voice data compressed package is made of 3 compress speech frames, the voice pressure in first voice data compressed package
Contracting frame number consecutively is 3,2,1；The compress speech frame that wherein number is 3 is this data really to be sent, and it is 2,1 to number
Compress speech frame be data transmitted by the continuous two voice data compressed packages sent before；Therefore, if do not occurred
If packet drop, the compress speech frame of number 2,1 is otiose, after the compress speech frame of number 2,1 is used only to
Packet is looked for during continuous packet loss.
Assuming that：Compress speech frame number consecutively in first voice data compressed package is 3,2,1；Second voice data
Compress speech frame number consecutively in compressed package is 4,3,2；Compress speech frame in third voice data compressed package is compiled successively
Number be 5,4,3；Next voice data compressed package and so on.
Following embodiment is to illustrate the situation of second voice data compression packet loss.
Step 1, receiving terminal receive the voice data compressed package according to natural number number consecutively；
The voice data compressed package that step 2, receiving terminal receive twice in succession is respectively first voice data compressed package
With third voice data compressed package, at this time receiving terminal calculate the number difference of two voice data compressed packages, number difference is
2, as transmission process is lost 1 voice data compressed package, is used to restore from third voice data compressed package from 2 at this time
The compress speech frame being connect in the compress speech frame of packet loss with the first compress speech frame starts to read, that is, reads number immediately
Compress speech frame, as number be 2 compress speech frame, due to only losing a packet, as long as so reading one voice pressure
Contracting frame.Such mode can restore the data content of loss.
If the number difference is more than 2, primary voice number after reading in the voice data compressed package received twice
According to whole compress speech frames in compressed package, the compress speech of number 2 and number 1 as in third voice data compressed package
Frame.
Due to the value of N can use it is big also can use small, when value is excessive, bandwidth availability ratio is just very low, and efficiency of transmission is just very slow,
When value is too small, the data of loss cannot restore as far as possible, cause communication quality low, however lead to an excess amount of test
Go out, when the value of N is 10, realize that efficiency of transmission is most fast while reduction rate is high.
Embodiment two
It is similar with embodiment one, by taking N is 10 as an example；
Compress speech frame number consecutively in first voice data compressed package is 10,9,8,7,6,5,4,3,2,1；
Compress speech frame number consecutively in second voice data compressed package is 11,10,9,8,7,6,5,4,3,2；
Compress speech frame number consecutively in third voice data compressed package is 12,11,10,9,8,7,6,5,4,3；
Compress speech frame number consecutively in 4th voice data compressed package is 13,12,11,10,9,8,7,6,5,4；
Compress speech frame number consecutively in 5th voice data compressed package is 14,13,12,11,10,9,8,7,6,5；
Compress speech frame number consecutively in 6th voice data compressed package is 15,14,13,12,11,10,9,8,7,
6；
Compress speech frame number consecutively in 7th voice data compressed package is 16,15,14,13,12,11,10,9,8,
7；
Next voice data compressed package and so on.
The present embodiment is second is that illustrate the situation of-five voice data compression packet loss of third.
Step 1, receiving terminal receive the voice data compressed package according to natural number number consecutively；
The voice data compressed package that step 2, receiving terminal receive twice in succession is respectively second voice data compressed package
With the 6th voice data compressed package, at this time receiving terminal calculate the number difference of two voice data compressed packages, number difference is
4, as transmission process is lost 3 voice data compressed packages, is used to restore from the 6th voice data compressed package from 9 at this time
The compress speech frame being connect in the compress speech frame of packet loss with the first compress speech frame starts to read, that is, reads number immediately
Compress speech frame, as number be 14 compress speech frame start, due to being lost 3 voice data compressed packages, thus after
Resume studies compress speech frame that number is taken to be 13 and compress speech frame that number is 12.However the voice for being 12 according to number
Condensed frame starts to play, and such mode can restore the data content of loss.
If the number difference is more than 10, primary voice number after reading in the voice data compressed package received twice
According to whole compress speech frames in compressed package, such as difference is 11, as loses 10 voice data compressed packages, reads work as at this time
All data in preceding voice data compressed package, although the content of also poor 1 voice data compressed package can not be read, due to language
The continuity of speech, when N values get sufficiently large, reduction degree can reach very high.
Due to the value of N can use it is big also can use small, when value is excessive, bandwidth availability ratio is just very low, and efficiency of transmission is just very slow,
When value is too small, the data of loss cannot restore as far as possible, cause communication quality low, however lead to an excess amount of test
Go out, when the value of N is 10, realize that efficiency of transmission is most fast while reduction rate is high.
Embodiment three
It is similar with embodiment one, by taking N is 15 as an example；
Compress speech frame number consecutively in first voice 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 voice 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 third voice 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 voice 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 voice data compressed package is 19,18,17,16,15,14,13,12,
11,10,9,8,7,6,5；
Next voice data compressed package and so on；
Compress speech frame number consecutively in tenth voice 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 17th voice data compressed package is 31,30,29,28,27,26,25,
24,23,22,21,20,19,18,17；
The present embodiment three is the situation for illustrating the 5th-nine voice data compression packet loss.
Step 1, receiving terminal receive the voice data compressed package according to natural number number consecutively；
The voice data compressed package that step 2, receiving terminal receive twice in succession is respectively the 4th voice data compressed package
With the tenth voice data compressed package, at this time receiving terminal calculate the number difference of two voice data compressed packages, number difference is
6, as transmission process is lost 5 voice data compressed packages, at this time from the tenth voice data compressed package from 14 for also
The compress speech frame being connect in the compress speech frame of former packet loss with the first compress speech frame starts to read, that is, reads number tightly
The compress speech frame connect, the compress speech frame that as number is 23 starts, due to being lost 5 voice data compressed packages, so
Continue to read the compress speech frame that number is 22,21,20,19.Then the compress speech frame for being 19 according to number starts to play, this
The mode of sample can restore the data content of loss.
If the number difference is more than 15, primary voice number after reading in the voice data compressed package received twice
According to whole compress speech frames in compressed package, such as the voice data compressed package received twice is respectively first voice data
Compressed package and the 17th voice data compressed package, this time difference value are 16, as lose 15 voice data compressed packages, read at this time
All data in current speech data compressed package, as number is 30,29,28,27,26,25,24,23,22,21,20,19,
18,17 compress speech frame, it is continuous due to language although the content of also poor 1 voice data compressed package can not be read
Property, when N values get sufficiently large, reduction degree can reach very high.
Due to the value of N can use it is big also can use small, when value is excessive, bandwidth availability ratio is just very low, and efficiency of transmission is just very slow,
When value is too small, the data of loss cannot restore as far as possible, cause communication quality low, however lead to an excess amount of test
Go out, when the value of N is 10, realize that efficiency of transmission is most fast while reduction rate is high.
In conclusion a kind of voice-optimizing transmission method provided by the invention and device, by sacrificing fractional transmission bandwidth,
The data content normally read containing the compressed package in each voice data compressed package, further includes the continuous number sent before
According to content, the number difference of voice data compressed package received twice by calculating connection, it can be determined that whether data occur
If packet loss situation without packet loss, only reads the data content that the compressed package is normally read, if packet loss, from current speech
The voice data compressed package of loss is restored in the continuous data content sent before compression data packet.It is passed generally for data
Defeated efficiency is that will not use above-mentioned this mode adequately using transmission bandwidth, and the technical solution adopted in the present invention master
If in order to reduce the probability of packet loss, by sacrificing fractional transmission bandwidth, the data of loss are restored as far as possible, ensure that voice is clear
Clear smoothness ensure that the quality of communication.In addition this programme is without storing data, and then can provide efficiency.
The foregoing is merely the embodiment of the present invention, are not intended to limit the scope of the invention, every to utilize this hair
The equivalents that bright specification and accompanying drawing content are made directly or indirectly are used in relevant technical field, similarly include
In the scope of patent protection of the present invention.
Claims (8)
1. a kind of voice-optimizing transmission method, which is characterized in that including：
Step 1 is received according to the voice data compressed package of natural number number consecutively；The voice data compressed package is by N number of voice
Condensed frame forms, respectively 1 voices for being used to restore packet loss for the first compress speech frame of voice data reading and N-1
Condensed frame；The N-1 are continuous voice data before the first compress speech frame for restoring the compress speech frame of packet loss；N's
Value is the integer more than 1；
Step 2, the number difference of voice data compressed package that receives twice in succession of calculating, if the number difference more than 1 and
Less than or equal to N, then it is used to restore packet loss from the N-1 in rear primary voice data compressed package successively according to number difference
Compress speech frame in the compress speech frame that is connect with the first compress speech frame start to read；If the number difference is more than N,
Whole compress speech frames after reading in the voice data compressed package received twice in primary voice data compressed package.
2. voice-optimizing transmission method according to claim 1, which is characterized in that the step 2 further includes：If the volume
Number difference is equal to 1, then after reading in the voice data compressed package received twice in primary voice data compressed package first
Compress speech frame.
3. voice-optimizing transmission method according to claim 1, which is characterized in that the voice data compressed package it is N number of
Compress speech frame uses natural number number consecutively.
4. voice-optimizing transmission method according to claim 1, which is characterized in that the value of the N is 10.
5. a kind of voice-optimizing transmitting device, which is characterized in that including receiving module and computing module；
The receiving module, for receiving the voice data compressed package according to natural number number consecutively；The voice data compression
Packet is made of N number of compress speech frame, and respectively 1 for the first compress speech frame of voice data reading and N-1 for also
The compress speech frame of former packet loss；The N-1 are continuous before the first compress speech frame for restoring the compress speech frame of packet loss
Voice data；The value of N is the integer more than 1；
The computing module includes computing unit, the first reading unit and the second reading unit；
The computing unit, for calculating the number difference of voice data compressed package received twice in succession；
First reading unit, if being more than 1 and less than or equal to N for the number difference, according to number difference successively from
The N-1 in primary voice data compressed package are used in the compress speech frame for restoring packet loss and the first compress speech frame afterwards
The compress speech frame of connection starts to read；
Second reading unit if being more than N for the number difference, reads the voice data compressed package received twice
In after whole compress speech frames in primary voice data compressed package.
6. voice-optimizing transmitting device according to claim 5, which is characterized in that the computing module further includes third reading
Take unit；
The third reading unit if being equal to 1 for the number difference, reads the voice data compressed package received twice
In after the first compress speech frame in primary voice data compressed package.
7. voice-optimizing transmitting device according to claim 5, which is characterized in that the voice data compressed package it is N number of
Compress speech frame uses natural number number consecutively.
8. voice-optimizing transmitting device according to claim 5, which is characterized in that the value of the N is 10.
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| 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 |
| CN112908346B (en) * | 2019-11-19 | 2023-04-25 | 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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