CN106209773A - The method that the sampling transmission of a kind of audio packet is recombinated again - Google Patents
The method that the sampling transmission of a kind of audio packet is recombinated again Download PDFInfo
- Publication number
- CN106209773A CN106209773A CN201610468238.6A CN201610468238A CN106209773A CN 106209773 A CN106209773 A CN 106209773A CN 201610468238 A CN201610468238 A CN 201610468238A CN 106209773 A CN106209773 A CN 106209773A
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- data
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- adjacent
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- 238000005070 sampling Methods 0.000 title claims abstract description 28
- 230000005540 biological transmission Effects 0.000 title claims abstract description 22
- 238000000034 method Methods 0.000 title claims abstract description 18
- 238000005194 fractionation Methods 0.000 claims 1
- 230000000903 blocking effect Effects 0.000 abstract description 2
- 230000000694 effects Effects 0.000 abstract description 2
- 230000006835 compression Effects 0.000 abstract 1
- 238000007906 compression Methods 0.000 abstract 1
- 230000006798 recombination Effects 0.000 abstract 1
- 238000005215 recombination Methods 0.000 abstract 1
- 238000010586 diagram Methods 0.000 description 2
- 230000002452 interceptive effect Effects 0.000 description 2
- 238000012986 modification Methods 0.000 description 2
- 230000004048 modification Effects 0.000 description 2
- 230000009286 beneficial effect Effects 0.000 description 1
- 239000000945 filler Substances 0.000 description 1
- 239000013589 supplement Substances 0.000 description 1
Classifications
-
- 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
- H04L65/60—Network streaming of media packets
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/002—Dynamic bit allocation
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L67/00—Network arrangements or protocols for supporting network services or applications
- H04L67/50—Network services
- H04L67/56—Provisioning of proxy services
- H04L67/565—Conversion or adaptation of application format or content
- H04L67/5651—Reducing the amount or size of exchanged application data
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L69/00—Network arrangements, protocols or services independent of the application payload and not provided for in the other groups of this subclass
- H04L69/16—Implementation or adaptation of Internet protocol [IP], of transmission control protocol [TCP] or of user datagram protocol [UDP]
- H04L69/161—Implementation details of TCP/IP or UDP/IP stack architecture; Specification of modified or new header fields
Landscapes
- Engineering & Computer Science (AREA)
- Signal Processing (AREA)
- Multimedia (AREA)
- Computer Networks & Wireless Communication (AREA)
- Computer Security & Cryptography (AREA)
- Computational Linguistics (AREA)
- Health & Medical Sciences (AREA)
- Audiology, Speech & Language Pathology (AREA)
- Human Computer Interaction (AREA)
- Physics & Mathematics (AREA)
- Acoustics & Sound (AREA)
- Data Exchanges In Wide-Area Networks (AREA)
Abstract
The invention discloses a kind of audio packet sampling and transmit recombination method again, by at collection terminal, each audio frequency pcm data cell is split by unit sampled point uniform intervals, splits into two groups of (or many groups) data cells, carry out compression coding, transmission respectively.By receiving terminal, data are decoded, if two groups of (or many groups) data cells all receive, then reassemble into complete audio data unit；If only receiving one group, then expand sampling number evidence, be allowed to there is identical sampling precision with collection terminal.This packet mode that this patent proposes can restore major part raw tone effect, thus reduce the probability blocking or losing after the voice data after only receiving a frame packet.And after receiving complete packet data, it is also possible to complete reduction tonequality.
Description
Technical field
The invention belongs to technical field of data transmission, especially carry out in the scene have interactive application to audio transmission
There is the system of high requirements.
Background technology
In current the Internet+society, people watch video by the Internet, live etc. become very frequently and
Usually.Real-time and fluency are required the highest, especially in interactive application scene (as regarded by the audio transmission on the Internet
Frequency call, video conference etc.), the card of audio frequency pauses and can allow very sensitive the perceiving of people.Traditional implementation is mono-for collecting pcm
After metadata, it is compressed coding and transmission, plays end and be decoded again playing.After having frame data to block or losing,
Have significantly card, show poor Consumer's Experience.
In consideration of it, exigence transmits the method recombinated again to solve prior art with the presence of the sampling of a kind of audio packet
Problem and risk.The present invention is just used to solve problems, and this packet mode that this patent proposes can only received
After voice data after a frame packet, restore major part raw tone effect, thus reduce the probability blocking or losing.And
After receiving complete packet data, it is also possible to complete reduction tonequality, farthest ensure customer experience.
Summary of the invention
It is an object of the invention to provide the method that the sampling transmission of a kind of audio packet is recombinated again, solve in prior art
The problems existed.
In order to realize the purpose of the present invention, the invention provides the method that the sampling transmission of a kind of audio packet is recombinated again, institute
The method of stating comprises the steps:
Step 1: in the data buffer storage that collection terminal is collected to CaptureBuffer.
Step 2: the CaptureBuffer that unit sampled splits, and odd bits sampled data is stored to
In CaptureBuffer1, even bit sampled data is stored in CaptureBuffer2.The two splits Buffer and all comprises phase
Sampled point with number.
Step 3: prepare two encoders, to the two CaptureBuffer1 and CaptureBuffer2 audio frequency number
Become EncoderBuffer1 and EncoderBuffer2 according to carrying out separately compressed encoding, more separately transmit.
Step 4: receiving terminal prepares two decoders, by the EncoderBuffer1 received or
EncoderBuffer2 is decoded into DecoderBuffer1 or DecoderBuffer2 respectively.
If 1. EncoderBuffer1 and EncoderBuffer2 all receives, then the two is decoded buffer and reassemble into
PlayBuffer, odd bits DecoderBuffer1 fill, and even bit DecoderBuffer2 fills.
If 2. EncoderBuffer1 or EncoderBuffer2 only receives one, then this is decoded
Buffer data are done and are expanded, and are saved in PlayBuffer.If that receive is EncoderBuffer1, then will
Adjacent two sampling numbers of DecoderBuffer1 are average according to doing, then are inserted into adjacent using these data as new sampled point
In the middle of two sampled points, last sampled point directly copies；If that receive is EncoderBuffer2, the one of foremost
Individual sampled point directly copies, more adjacent for DecoderBuffer2 two sampling numbers are average according to doing, using these data as
New sampled point is inserted in the middle of adjacent two sampled point.PlayBuffer after restructuring comprises adopting of original equal number
Sampling point.
Step 5: deliver to the PlayBuffer of generation play in player.
The present invention, compared with prior art, by using multilink transmission voice data, in obstructed application scenarios
Faster transfer rate or more stable transmission quality can be provided.Compared with prior art, factor data can be preferably avoided to hinder
The audio card that cause such as plug are paused problem, it is thus possible to preferably improve Consumer's Experience.
Accompanying drawing explanation
Fig. 1 is the method flow diagram of the present invention.
Fig. 2 is example 1 schematic diagram of the present invention.
Instantiation mode
In order to make the purpose of the present invention, technical scheme and beneficial effect are clearer, below in conjunction with example, enter the present invention
Row further describes.Should be understood to that instantiation described herein, only in order to explain the present invention, is not used to limit
Protection scope of the present invention processed.
As shown in Fig. 1, the invention provides the method that the sampling transmission of a kind of audio packet is recombinated again, described method includes
Following steps:
Step S101, in the data buffer storage collected by collection terminal to CaptureBuffer.Assume to gather to comprise 256 every time
Sampled point.
Step S102, the CaptureBuffer that unit sampled splits, and odd bits sampled data is stored to
In CaptureBuffer1, even bit sampled data is stored in CaptureBuffer2.The two splits Buffer and all comprises 128
Individual sampled point.
Step S103, prepares two encoders (sample rate be 8K, sampling resolution 16, monophonic), to the two
CaptureBuffer1 with CaptureBuffer2 voice data carry out separately compressed encoding become EncoderBuffer1 and
EncoderBuffer2, more separately transmit.
Step S104, receiving terminal prepares two decoders (sample rate be 8K, sampling resolution 16, monophonic), will receive
EncoderBuffer1 or EncoderBuffer2 be decoded into DecoderBuffer1 or DecoderBuffer2 respectively, respectively wrap
Containing 128 sampled points.
If 1. EncoderBuffer1 and EncoderBuffer2 all receives, then the two is decoded buffer and reassemble into
PlayBuffer, odd bits DecoderBuffer1 fill, and even bit DecoderBuffer2 fills.
If 2. EncoderBuffer1 or EncoderBuffer2 only receives one, then this is decoded
Buffer data are done and are expanded, and are saved in PlayBuffer.If that receive is EncoderBuffer1, then will
Adjacent two sampling numbers of DecoderBuffer1 are average according to doing, then are inserted into adjacent using these data as new sampled point
In the middle of two sampled points, last sampled point directly copies；If that receive is EncoderBuffer2, the one of foremost
Individual sampled point directly copies, more adjacent for DecoderBuffer2 two sampling numbers are average according to doing, using these data as
New sampled point is inserted in the middle of adjacent two sampled point.PlayBuffer after restructuring comprises original 256 sampled point.
Step S105, delivers to play in player (16K sample rate, sampling resolution 16, monophone by the PlayBuffer of generation
Road).
The above is only the optimal way of the present invention, it is noted that be not limited to sampled point divides two groups of transmission, as long as
Be use packet transmission strategy be all protection scope of the present invention.Should also be pointed out that the ordinary skill people for the art
For Yuan, under the premise without departing from the principles of the invention, it is also possible to make some improvements and modifications, these improvements and modifications also should
It is considered as protection scope of the present invention.
As in figure 2 it is shown, instantiation.
The method that a kind of audio packet sampling transmission of the present invention introduced below is recombinated again.
Example 1
As illustrated in fig. 2, it is assumed that application scenarios is a Web conference the highest to requirements of real time, wanting according to step S101
Ask, in the data buffer storage that collection terminal is collected to CaptureBuffer.Assume to gather to comprise 256 sampled points every time.
According to step S102, the CaptureBuffer that unit sampled splits, and odd bits sampled data is stored to
In CaptureBuffer1, even bit sampled data is stored in CaptureBuffer2.The two splits Buffer and all comprises 128
Individual sampled point.
According to step S103, prepare two encoders (sample rate be 8K, sampling resolution 16, monophonic), to the two
CaptureBuffer1 with CaptureBuffer2 voice data carry out separately compressed encoding become EncoderBuffer1 and
EncoderBuffer2, wherein a road uses the mode of TCP to transmit, and an other road uses the mode of UDP to transmit.
According to step S104, receiving terminal prepares two decoders (sample rate be 8K, sampling resolution 16, monophonic), respectively
From the line-receiving data of TCP and UDP.EncoderBuffer1 or EncoderBuffer2 received is decoded into respectively
DecoderBuffer1 or DecoderBuffer2, respectively comprises 128 sampled points.Assume in this example that by TCP line transmission
Data all have received in EncoderBuffer1, and the loss of data of UDP transmission does not arrives EncoderBuffer2.This example is with very
Numerical digit 1,3,13,15 is by way of example it is assumed that the sample magnitude of their correspondence is 46,38,21,69.By in EncoderBuffer1
Data decoding after put in DecodeBuffer1, take the new sampled point in the meansigma methods seat of adjacent two points be inserted into this two
The centre of individual neighbouring sample point, last sampled point direct copying.As in figure 2 it is shown, by odd bits 1 and odd bits 3 in this example
Sample magnitude do averagely, drawn (the 46+38)/2=42 filling numerical value as even bit 2, same, odd bits 13 and strange
The sampled value of several 15 is done averagely, show that (21+69)/2=45 finishes the filler of even bit 14.By the method, by TCP this
128 odd bits sampled points that road is transmitted through, supplement as 256 original sampled points and put in PlayBuffer.
According to step S105, the PlayBuffer of generation is delivered in player play (16K sample rate, sampling resolution 16,
Monophonic).
Claims (3)
1. the method that an audio packet sampling transmission is recombinated again, it is characterised in that comprise the steps: step 1: by collection terminal
The data buffer storage collected in CaptureBuffer, step 2: the CaptureBuffer that unit sampled splits, very
Numerical digit sampled data is stored in CaptureBuffer1, and even bit sampled data is stored in CaptureBuffer2, and the two is torn open
Buffer is divided all to comprise the sampled point of same number, step 3: prepare two encoders, to the two CaptureBuffer1
Carry out separately compressed encoding with CaptureBuffer2 voice data and become EncoderBuffer1 and EncoderBuffer2, then enter
Row separately transmission, step 4: receiving terminal prepares two decoders, by the EncoderBuffer1 received or
EncoderBuffer2 is decoded into DecoderBuffer1 or DecoderBuffer2 respectively, if EncoderBuffer1 and
EncoderBuffer2 all receives, then the two decodes buffer and reassembles into PlayBuffer, and odd bits is used
DecoderBuffer1 fill, even bit with DecoderBuffer2 fill, if 2. EncoderBuffer1 or
EncoderBuffer2 only receives one, then these decoded buffer data done and expand, and be saved in
In PlayBuffer, if that receive is EncoderBuffer1, then by adjacent for DecoderBuffer1 two sampling number evidences
Doing average, then these data be inserted in the middle of adjacent two sampled point as new sampled point, last sampled point is straight
Connect and copy；If that receive is EncoderBuffer2, a sampled point of foremost directly copies, then will
Adjacent two sampling numbers of DecoderBuffer2 are average according to doing, and these data are inserted into adjacent two as new sampled point
In the middle of individual sampled point, the PlayBuffer after restructuring comprises the sampled point of original equal number, step 5: by generate
PlayBuffer delivers to play in player.
2. the method that an audio packet according to claim 1 sampling transmission is recombinated again, it is characterised in that in step 2
Fractionation mode be not limited to 2 groups, sampled point is all its feature by the strategy that packet is transmitted.
3. the method that an audio packet according to claim 1 sampling transmission is recombinated again, it is characterised in that in step 4
In method 2 when data fail complete transmission, the expansion of data uses adjacent two sampling numbers average according to doing, then by this number
According to being inserted in the middle of adjacent two sampled point as new sampled point, finally or the junior one sampled point directly makees the side copied
Method ensures that the sample rate after restructuring is identical in crude sampling rate.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201610468238.6A CN106209773A (en) | 2016-06-24 | 2016-06-24 | The method that the sampling transmission of a kind of audio packet is recombinated again |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201610468238.6A CN106209773A (en) | 2016-06-24 | 2016-06-24 | The method that the sampling transmission of a kind of audio packet is recombinated again |
Publications (1)
| Publication Number | Publication Date |
|---|---|
| CN106209773A true CN106209773A (en) | 2016-12-07 |
Family
ID=57461033
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CN201610468238.6A Pending CN106209773A (en) | 2016-06-24 | 2016-06-24 | The method that the sampling transmission of a kind of audio packet is recombinated again |
Country Status (1)
| Country | Link |
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| CN (1) | CN106209773A (en) |
Cited By (2)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN109584889A (en) * | 2018-12-28 | 2019-04-05 | 秒针信息技术有限公司 | Audio frequency transmission method and device and storage medium |
| CN113472944A (en) * | 2021-08-05 | 2021-10-01 | 苏州欧清电子有限公司 | Voice self-adaptive processing method, device, equipment and storage medium of intelligent terminal |
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| CN101313588A (en) * | 2005-09-27 | 2008-11-26 | 高通股份有限公司 | Scalability techniques based on content information |
| US20110119565A1 (en) * | 2009-11-19 | 2011-05-19 | Gemtek Technology Co., Ltd. | Multi-stream voice transmission system and method, and playout scheduling module |
| US20130144630A1 (en) * | 2002-09-04 | 2013-06-06 | Microsoft Corporation | Multi-channel audio encoding and decoding |
| CN103325373A (en) * | 2012-03-23 | 2013-09-25 | 杜比实验室特许公司 | Method and equipment for transmitting and receiving sound signal |
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2016
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| US20130144630A1 (en) * | 2002-09-04 | 2013-06-06 | Microsoft Corporation | Multi-channel audio encoding and decoding |
| CN101313588A (en) * | 2005-09-27 | 2008-11-26 | 高通股份有限公司 | Scalability techniques based on content information |
| US20110119565A1 (en) * | 2009-11-19 | 2011-05-19 | Gemtek Technology Co., Ltd. | Multi-stream voice transmission system and method, and playout scheduling module |
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Cited By (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN109584889A (en) * | 2018-12-28 | 2019-04-05 | 秒针信息技术有限公司 | Audio frequency transmission method and device and storage medium |
| CN109584889B (en) * | 2018-12-28 | 2021-07-20 | 秒针信息技术有限公司 | Audio transmission method and device and storage medium |
| CN113472944A (en) * | 2021-08-05 | 2021-10-01 | 苏州欧清电子有限公司 | Voice self-adaptive processing method, device, equipment and storage medium of intelligent terminal |
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| WD01 | Invention patent application deemed withdrawn after publication |
Application publication date: 20161207 |
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