CN105337897B - A kind of audio PTT synchronous transmission system based on RTP message - Google Patents
A kind of audio PTT synchronous transmission system based on RTP message Download PDFInfo
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- CN105337897B CN105337897B CN201510741106.1A CN201510741106A CN105337897B CN 105337897 B CN105337897 B CN 105337897B CN 201510741106 A CN201510741106 A CN 201510741106A CN 105337897 B CN105337897 B CN 105337897B
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L49/00—Packet switching elements
- H04L49/60—Software-defined switches
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L49/00—Packet switching elements
- H04L49/90—Buffering arrangements
- H04L49/9057—Arrangements for supporting packet reassembly or resequencing
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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
- H04L65/60—Network streaming of media packets
- H04L65/65—Network streaming protocols, e.g. real-time transport protocol [RTP] or real-time control protocol [RTCP]
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- Computer Networks & Wireless Communication (AREA)
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- Data Exchanges In Wide-Area Networks (AREA)
Abstract
The invention discloses a kind of audio PTT synchronous transmission system based on RTP message, including DSP signal acquisition module, CPU and DSP output control module, the DSP signal acquisition module and CPU, the CPU is connect with the DSP output control module, wherein: the DSP signal acquisition module is used to acquire the control audio PTT and voice audio two parts of signals in radio station, digitized processing is carried out to control audio PTT and voice audio, the control audio PTT at same time point and voice audio information are merged；The CPU carries out RTP packing for reading the pooling information, and by the pooling information, is then parsed again to the RTP by network transmission；The DSP output control module is used to control the synchronizing information output of control audio PTT and voice audio after parsing.The present invention realizes sound audio signal and controls the synchronous transfer of audio PTT signal, while solving the problems, such as network transfer delay, improves network transmission quality.
Description
Technical field
The present invention relates to speech processes field, specifically a kind of audio PTT synchronous transmission system based on RTP message.
Background technique
General radio station control all includes control audio PTT and voice audio two parts, its word of existing most of radio station
Sound audio and control audio PTT are to separate transmission, can thus there is two nonsynchronous problems of signal.
With the application of Softswitch technology, these analog signals need to carry out network transmission, right in military industry field application
What the synchronous requirement of two signals was strict, if the stationary problem of two signals cannot be handled well, it will serious
The integrality of signal and the accuracy of data transmission are influenced, so in the voice audio of radio present and control audio PTT transmission
Synchronization and delay problem it is urgently to be resolved.
Summary of the invention
Object of the present invention is to overcome the deficiencies of the prior art and provide a kind of audio PTT synchronous transfer system based on RTP message
System can make to control audio PTT and voice audio signal synchronizes network transmission, guarantee transmitting terminal to signal between receiving end
The integrality of transmission.
The technical scheme of the present invention is realized as follows: including DSP signal acquisition module, the first CPU, the 2nd CPU and DSP
Output control module, the DSP signal acquisition module are connect with the first CPU, and the first CPU is connect with the 2nd CPU, institute
The 2nd CPU is stated to connect with the DSP output control module, in which:
The DSP signal acquisition module is used to acquire the control audio PTT and voice audio two parts of signals in radio station, to control
Audio PTT processed and voice audio carry out digitized processing, and the control audio PTT at same time point and voice audio information are distinguished
Generation pooling information is merged after independent preservation；
First CPU carries out RTP packing for reading the pooling information, and by the pooling information, then passes through
Network transmission gives the 2nd CPU；
2nd CPU generates mutually independent control audio PTT and words for parsing to the RTP data packet
Sound audio；
Specifically, the RTP is packaged and RTP resolve packet is referred to as Real-time Transport Protocol, and the Real-time Transport Protocol is opened by IETF
Hair, establishment prestige RFC official documentation in 1996 is for VoIP, video, the agreement of the real-time medias such as voice transmission.RTP is to hand over
The data with real-time characteristic such as mutual formula audio, video provide transmission service end to end.On ip networks, usually in UDP
On run Real-time Transport Protocol.
Heretofore described RTP is packaged and the concrete mode of RTP resolve packet is as follows:
(1) single NAL is packaged
H.264NALU unit is often made of [start code] [NALU header] [NALU payload] three parts,
Middle start code is used for the beginning of mark one NALU unit, it is necessary to be " 00000001 " either " 000001 ", when packing
Remove opening code, other data are bundled to RTP packet can.
(2) fragment is packaged
Since 1500 bytes are the upper limits of the length of IP datagram, the datagram header of 20 bytes, 1480 words are removed
Section is used to store UDP datagram.So we must beat its fragment when the byte number in a frame is more than this numerical value
Packet.And UDP during transmission also will be by packet header expense, so the maximum number of byte of RTP packet is positioned 1400 bytes.
The process analysis of packing and unpacking:
It is packaged:
It is described in detail when fragment:
1. the FU indicator of first FU-A packet is so arranged: in the F in F=NALU, NRI=NALU
NRI, the Type in Type=28FU header:S=1, E=0, R=0, Type=NALU；
2. the FU indicator of intermediate FU-A packet is so arranged: in the F in F=NALU, NRI=NALU
NRI, the Type in Type=28FU header:S=0, E=0, R=0, Type=NALU；
3. the FU indicator of tail FU-A packet is so arranged: in the F in F=NALU, NRI=NALU
Type in NRI, Type=28FU header:S=0, E=1, R=0, Type=NALU.
It unpacks:
The fragmented code realization classified is analyzed when we unpack for RTP below:
Byte startBit=(byte) (recbuf [13] &0x80)；Byte endBit=(byte) (recbuf
[13]&0x40)；
1. this packet is the first packet of fragment if startBit==-128.
NalBuf [4]=(byte) ((recbuf [12] &0xE0)+(recbuf [13] &0x1F))；This sentence is for rebuilding group
Close NAL unit type.
2. if (startBit==0) && (endBit==0), this packet are the middle sections of fragment.
3. this packet is fragment tail portion if endBit==64.
It is clear when classifying, so that it may which that corresponding processing is done to each section.
The synchronizing information that the DSP output control module is used to control control audio PTT and voice audio after parsing is defeated
Out.
Specifically, the DSP signal acquisition module is connect using HPI bus interface with the first CPU.
Further, the 2nd CPU is connect using HPI bus interface with the DSP output control module.
Further, when the RTP is parsed by network transmission, skill is buffered using jitterbuffer network packet
Art, for solving the delay problem of RTP network transmission.
The variation of data packet arrival time, referred to as shakes, shake can due to network congestion, timing wander or routing change and
It generates.Using jitterbuffer network packet buffer technology, i.e., wobble buffer is put in the receiving end of voice connection, by shaking
The data packet that it can operatively be delayed to reach known to the function of buffer, in this way, terminal user will experience one clearly
Clear, without the connection of what audio distortions.
Specifically, the DSP signal acquisition module includes thick film circuit, FPGA and dsp chip, the voice audio signal
It is converted into the digital signal that A restrains format by thick film circuit, then into the MCBSP of the dsp chip；The control sound
Frequency PTT is AD converted by FPGA, then into the MCBSP of the dsp chip；
The MCBSP of the dsp chip uses DMA transfer mechanism, by collected voice audio signal and control audio PTT
Signal is stored in the corresponding storage address of external RAM, and the first CPU is notified to be read out, and completes adopting for entire signal
Collection.
Specifically, the DSP output control module also includes thick film circuit, FPGA and dsp chip, the MCBSP of dsp chip
The control audio PTT data after parsing are written in the FPGA by being time-multiplexed, the FPGA carries out DA digital-to-analogue to it and turns
Externally output low and high level control is changed, realizes the control of control audio PTT；A after parsing is restrained format by the MCBSP of dsp chip
Spoken audio data be written in the thick film circuit by time division multiplexing, the thick film circuit is externally defeated by DA analog-to-digital conversion
Voice audio signals out.
Further, the dsp chip uses TMS320C6418 chip.
Further, the control flow of this system is as follows:
Signal acquisition and data merge: being carried out by the DSP signal acquisition module to control audio PTT and voice audio
Digitized processing, and life is merged after the control audio PTT at same time point and voice audio information are independently saved
At pooling information；
RTP is packaged: reading the pooling information by the first CPU and the pooling information is carried out RTP packing；
Parsing RTP data packet: the information after packing is by being conveyed to the 2nd CPU, the 2nd CPU after network transmission
The RTP data packet is parsed, and the RTP package informatin is parsed into mutual independent control audio PTT and speech again
Audio；
Load data output: by the DSP output module to after the parsing mutual independent control audio PTT with
Voice audio synchronizes output, and is received by controlled radio station is synchronous.
The beneficial effects of the present invention are: the present invention carries out voice audio signal and control audio PTT using DSP acquisition module
The acquisition of signal, and a RTP data packet is merged into after two digital signals are independently saved, it will be in RTP data packet
Pure load expansion is twice, and carries out the reduction of point data at the same time simultaneously after network transmission, then by DSP output control module
Output realizes sound audio signal and controls the synchronous transfer of audio PTT signal；The present invention additionally uses simultaneously
Jitterbuffer network packet buffer technology solves the delay problem of RTP network transmission, improves network transmission quality.
Detailed description of the invention
Fig. 1 is a kind of overall structure functional block diagram of the audio PTT synchronous transmission system based on RTP message of the present invention；
Fig. 2 is the connection figure of DSP signal acquisition module and the first CPU in the present invention being connect by HPI bus interface；
Fig. 3 is the signal acquisition functional block diagram of DSP signal acquisition module of the present invention；
Fig. 4 is DSP output control module output principle block diagram of the present invention；
Fig. 5 is a kind of control flow chart of the audio PTT synchronous transmission system based on RTP message of the present invention.
Specific embodiment
Specific embodiments of the present invention will be further explained with reference to the accompanying drawing:
As shown in Figure 1, a kind of audio PTT synchronous transmission system based on RTP message, including DSP signal acquisition module,
One CPU, the 2nd CPU and DSP output control module, the DSP signal acquisition module are connect with the first CPU, and described first
CPU is connect with the 2nd CPU, and the 2nd CPU is connect with the DSP output control module, in which: the DSP signal acquisition mould
Block is used to acquire the control audio PTT and voice audio two parts of signals in radio station, counts to control audio PTT and voice audio
Wordization processing, carries out being merged into merging after the control audio PTT at same time point and voice audio information are independently saved
Information；First CPU carries out RTP packing for reading the pooling information, and by the pooling information, then passes through network
It is transferred to the 2nd CPU, then the RTP data packet is parsed by the 2nd CPU, generates mutually independent control audio
PTT and voice audio when the RTP data packet is parsed by network transmission, are buffered using jitterbuffer network packet
Technology, for solving the problems, such as RTP network transfer delay；The DSP output control module is used to control the mutual independence after parsing
Control audio PTT and voice audio synchronizing information output.
In practical applications, above procedure can be explained are as follows: 8 bits are controlled audio PTT and 8 bit voice digital audios
Signal merges into the digital signal of 16 bits, is then put into RTP packet, and two signals still may be used in 16 bit entire in this way
With reduction, output end is arrived and has been then reduced into two independent signals respectively again, whole process guarantees two signal digital collections
It can achieve with reduction synchronous.It is that independent save controls audio in RTP packet in transmission process although being a RTP packet
PTT and voice audio information, it may be implemented completely to split and reduction.Certainly, according to specific practical application, the control audio
PTT and voice audio are not limited solely to 8 bits.
The DSP signal acquisition module is connect using HPI bus interface with the first CPU, is connect by the HPI bus
Mouthful, the first CPU can access the memory space of DSP.The connection of HPI bus interface is as shown in Figure 2.Likewise, described
Two CPU are connect using HPI bus interface with the DSP output control module.
Specifically, the RTP is packaged and RTP resolve packet is referred to as Real-time Transport Protocol, and the Real-time Transport Protocol is opened by IETF
Hair, establishment prestige RFC official documentation in 1996 is for VoIP, video, the agreement of the real-time medias such as voice transmission.RTP is to hand over
The data with real-time characteristic such as mutual formula audio, video provide transmission service end to end.On ip networks, usually in UDP
On run Real-time Transport Protocol.
Heretofore described RTP is packaged and the concrete mode of RTP resolve packet is as follows:
(1) single NAL is packaged
H.264NALU unit is often made of [start code] [NALU header] [NALU payload] three parts,
Middle start code is used for the beginning of mark one NALU unit, it is necessary to be " 00000001 " either " 000001 ", when packing
Remove opening code, other data are bundled to RTP packet can.
(2) fragment is packaged
Since 1500 bytes are the upper limits of the length of IP datagram, the datagram header of 20 bytes, 1480 words are removed
Section is used to store UDP datagram.So we must beat its fragment when the byte number in a frame is more than this numerical value
Packet.And UDP during transmission also will be by packet header expense, so the maximum number of byte of RTP packet is positioned 1400 bytes.
The process analysis of packing and unpacking:
It is packaged:
It is described in detail when fragment:
1. the FU indicator of first FU-A packet is so arranged: in the F in F=NALU, NRI=NALU
NRI, the Type in Type=28FU header:S=1, E=0, R=0, Type=NALU；
2. the FU indicator of intermediate FU-A packet is so arranged: in the F in F=NALU, NRI=NALU
NRI, the Type in Type=28FU header:S=0, E=0, R=0, Type=NALU；
3. the FU indicator of tail FU-A packet is so arranged: in the F in F=NALU, NRI=NALU
Type in NRI, Type=28FU header:S=0, E=1, R=0, Type=NALU.
It unpacks:
The fragmented code realization classified is analyzed when we unpack for RTP below:
Byte startBit=(byte) (recbuf [13] &0x80)；Byte endBit=(byte) (recbuf
[13]&0x40)；
1. this packet is the first packet of fragment if startBit==-128.
NalBuf [4]=(byte) ((recbuf [12] &0xE0)+(recbuf [13] &0x1F))；This sentence is for rebuilding group
Close NAL unit type.
2. if (startBit==0) && (endBit==0), this packet are the middle sections of fragment.
3. this packet is fragment tail portion if endBit==64.
It is clear when classifying, so that it may which that corresponding processing is done to each section.
Further, the jitterbuffer is wobble buffer, and in voice over IP (VoIP), shake is slow
Rushing device is a shared data area, and in this data area, every one section of uniform interval, voice packet can be collected,
It stores and is dealt into speech processor.
The variation of data packet arrival time, referred to as shakes, shake can due to network congestion, timing wander or routing change and
It generates.Using jitterbuffer network packet buffer technology, i.e., wobble buffer is put in the receiving end of voice connection, by shaking
The data packet that it can operatively be delayed to reach known to the function of buffer, in this way, terminal user will experience one clearly
Clear, without the connection of what audio distortions.
As shown in figure 3, the DSP signal acquisition module includes thick film circuit, FPGA and dsp chip, the voice audio
Signal is converted into the digital signal that A restrains format by thick film circuit, then occupies a time slot and enters the dsp chip
In MCBSP；The control audio PTT is AD converted by FPGA, is then occupied a time slot and is entered the dsp chip
In MCBSP；The MCBSP of the dsp chip uses DMA transfer mechanism, by collected voice audio signal and control audio PTT
Signal is stored in the corresponding storage address of external RAM, and the first CPU is notified to be read out, and completes adopting for entire signal
Collection.Specifically, the dsp chip uses TMS320C6418 chip.
Above-mentioned is the DSP signal acquisition process in radio station all the way, multichannel radio station signal acquisition can equally be introduced described
The MCBSP of dsp chip is handled, for accessing the bandwidth of MCBSP, when can access 256 for 64K transmission if it is 8M
Gap, the corresponding time slot of an audio, that is, can theoretically access 128 radio station.
The payload of voice transmission is the data of 10ms usually in RTP voice transfer.Audio data passes through AD conversion
Become afterwards A rule format (8bit data) enter in the MCBSP of the dsp chip, dsp chip by MCBSP according to audio
The sample rate of 8k, as soon as second 8000 points of acquisition, then 10ms will acquire 80 pcm word sections, that is, all the way audio in 10ms
The interior data that will generate 80 bytes.According to general voice transfer, the payload in such packet RTP data packet is exactly 80
A byte is transmitted.And the present invention is to have the control audio PTT and voice audio two-way audio data altogether 160 bytes
Left and right one RTP payload transmitted, be equivalent to and carried out two paths of data merging, and receiving end to a bag data also
Original is also the information put at the same time at two-way audio, all effectively to guarantee synchronism.
Wherein, the full name of the RTP is Real-time Transport Protocol (real-time transport protocol).It is IETF
The standard proposed, corresponding RFC document is RFC3550 (RFC1889 is its expired version).RTP is the upper end Internet
It is synchronous with stream that real-time Transmission to end provides temporal information.RTP implementor need to first encapsulate data into when sending RTP data
RTP packet, and RTP data packet is being received, it needs to extract data from RTP packet.
Shown in the following table of the header format of RTP:
18 wherein are set by load type, i.e., G.729 format, payload are spliced 160 bytes.It is other
Information is generated by the library ORTP automatic packaging.
The length in the domain timestamp (Timestamp) is 32 bytes.First character section adopts in its reflection RTP data packets
The sample moment (time).Receiving end can use this timestamp to remove the shake of the packet as caused by network, and connect
Receiving end provides synchronizing function to play.
Since network can not be highly desirable, and consumption when needing to handle to data packet sequencing, to obtain the data to have sorted
The time interval of packet is.Invention also uses jitterbuffer network packet buffer technology, can will receive
Data pack buffer gets up, and then according to the packaging information of data packet (such as packet serial number and time stamp), out-of-order packet is resequenced,
Finally the data packet resequenced is read out.It can effectively be eliminated using jitterbuffer network packet buffer technology
Delay variation.
Payload is parsed after the DSP output control module receives RTP packet as receiving end, by payload
160 bytes are divided into two parts up and down, correspond respectively to the voice audio and control audio PTT data.The DSP output
The output process of control module is the inverse process of the DSP signal acquisition module in process.
Specifically, the DSP output control module also includes thick film circuit, FPGA and dsp chip, the dsp chip
Using TMS320C6418 chip.It is wherein described after the DSP output control module receives the payload of RTP Packet analyzing
The control audio PTT data after parsing are written in the FPGA by time division multiplexing by the MCBSP of dsp chip, and described FPGA pairs
It carries out DA digital-to-analogue conversion and externally exports low and high level control, realizes the control of control audio PTT；The MCBSP of the dsp chip
The spoken audio data of A rule format after parsing is written in the thick film circuit by time division multiplexing, the thick film circuit is logical
It crosses DA analog-to-digital conversion and externally exports voice audio signals.
As shown in figure 5, further, the control flow of present system is as follows:
Signal acquisition and data merge: being carried out by the DSP signal acquisition module to control audio PTT and voice audio
Digitized processing, and the control audio PTT at same time point and voice audio information are independently saved and merge generation
Pooling information；
RTP is packaged: the pooling information read by the first CPU and the pooling information is subjected to RTP packing, and
The 2nd CPU is given by network transmission；
Parse RTP data packet: the 2nd CPU parses the RTP data packet, and the RTP data packet is believed
Breath is parsed into mutual independent control audio PTT and voice audio again；
Load data output: by the DSP output module to after the parsing mutual independent control audio PTT with
Voice audio synchronizes output, and is received by controlled radio station is synchronous.
According to the disclosure and teachings of the above specification, those skilled in the art in the invention can also be to above-mentioned embodiment party
Formula is changed and is modified.Therefore, the invention is not limited to the specific embodiments disclosed and described above, to the one of invention
A little modifications and changes should also be as falling into the scope of the claims of the present invention.In addition, although being used in this specification
Some specific terms, these terms are merely for convenience of description, does not limit the present invention in any way.
Claims (7)
1. a kind of audio PTT synchronous transmission system based on RTP message, which is characterized in that including DSP signal acquisition module,
One CPU, the 2nd CPU and DSP output control module, the DSP signal acquisition module are connect with the first CPU, and described first
CPU is connect with the 2nd CPU, and the 2nd CPU is connect with the DSP output control module, in which:
The DSP signal acquisition module is used to acquire the control audio PTT and voice audio two parts of signals in radio station, to control sound
Frequency PTT and voice audio carry out digitized processing, independently by the control audio PTT at same time point and voice audio information
Generation pooling information is merged after preservation；
First CPU carries out RTP packing for reading the pooling information, and by the pooling information, then passes through network
It is transferred to the 2nd CPU；
2nd CPU generates mutually independent control audio PTT and speech sound for parsing to the RTP data packet
Frequency information；
The DSP output control module is used to control the information of mutually independent control audio PTT and voice audio after parsing
Synchronism output；
The DSP signal acquisition module includes thick film circuit, FPGA and dsp chip, and the voice audio signal passes through thick film electricity
Road is converted into the digital signal of A rule format, then into the MCBSP of the dsp chip；The control audio PTT passes through
FPGA is AD converted, then into the MCBSP of the dsp chip；The MCBSP of the dsp chip uses DMA transfer machine
Collected voice audio signal and control audio PTT signal are stored in the corresponding storage address of external RAM, and lead to by system
Know that the first CPU is read out, completes the acquisition of entire signal.
2. a kind of audio PTT synchronous transmission system based on RTP message as described in claim 1, which is characterized in that described
DSP signal acquisition module is connect using HPI bus interface with the first CPU.
3. a kind of audio PTT synchronous transmission system based on RTP message as described in claim 1, which is characterized in that described the
Two CPU are connect using HPI bus interface with the DSP output control module.
4. a kind of audio PTT synchronous transmission system based on RTP message as described in claim 1, which is characterized in that described
When RTP is parsed by network transmission, using jitterbuffer network packet buffer technology, for solving RTP network transmission
Delay problem.
5. a kind of audio PTT synchronous transmission system based on RTP message as described in claim 1, which is characterized in that described
DSP output control module also includes thick film circuit, FPGA and dsp chip, and the MCBSP of the dsp chip will by time division multiplexing
Control audio PTT data after parsing are written in the FPGA, and the FPGA carries out DA digital-to-analogue to the control audio PTT data
Conversion externally exports low and high level control, realizes the control of control audio PTT；The MCBSP of the dsp chip will be after parsing
Spoken audio data is written in the thick film circuit by time division multiplexing, and the thick film circuit is externally defeated by DA analog-to-digital conversion
Voice audio signals out.
6. a kind of audio PTT synchronous transmission system based on RTP message as claimed in claim 1 or 5, which is characterized in that institute
Dsp chip is stated using TMS320C6418 chip.
7. a kind of audio PTT synchronous transmission system based on RTP message as claimed in claim 1 or 5, which is characterized in that its
Control flow is as follows:
Signal acquisition and data merge: carrying out number to control audio PTT and voice audio by the DSP signal acquisition module
Change processing, and merge generation after the control audio PTT at same time point and voice audio information are independently saved and close
And information；
RTP is packaged: reading the pooling information by the first CPU and the pooling information is carried out RTP packing；
Parse RTP data packet: the information after packing is by network transmission to the 2nd CPU, and the 2nd CPU is to the RTP data
Packet is parsed, and the RTP packet information is parsed into mutually independent control audio PTT and voice audio again；Load
Data output: by the DSP output module to after the parsing mutually independent control audio PTT and voice audio carry out
Synchronism output, and received by controlled radio station is synchronous.
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| CN114285513B (en) * | 2021-11-22 | 2023-10-27 | 杭州当虹科技股份有限公司 | Delay device and method for supporting long-time delay of lossless IP signal |
Citations (5)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| JP2004185680A (en) * | 2002-11-29 | 2004-07-02 | Casio Comput Co Ltd | Reproduction control device and reproduction control processing program |
| CN101188525A (en) * | 2007-11-27 | 2008-05-28 | 华为技术有限公司 | A processing method and device for voice stream |
| CN101490739A (en) * | 2006-07-14 | 2009-07-22 | 高通股份有限公司 | Improved methods and apparatus for delivering audio information |
| CN101827242A (en) * | 2010-05-10 | 2010-09-08 | 南京邮电大学 | Method for realizing video phone system based on IPTV set-top box |
| CN103188403A (en) * | 2011-12-30 | 2013-07-03 | 迈普通信技术股份有限公司 | Voice gateway online monitoring method |
-
2015
- 2015-10-31 CN CN201510741106.1A patent/CN105337897B/en active Active
Patent Citations (5)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| JP2004185680A (en) * | 2002-11-29 | 2004-07-02 | Casio Comput Co Ltd | Reproduction control device and reproduction control processing program |
| CN101490739A (en) * | 2006-07-14 | 2009-07-22 | 高通股份有限公司 | Improved methods and apparatus for delivering audio information |
| CN101188525A (en) * | 2007-11-27 | 2008-05-28 | 华为技术有限公司 | A processing method and device for voice stream |
| CN101827242A (en) * | 2010-05-10 | 2010-09-08 | 南京邮电大学 | Method for realizing video phone system based on IPTV set-top box |
| CN103188403A (en) * | 2011-12-30 | 2013-07-03 | 迈普通信技术股份有限公司 | Voice gateway online monitoring method |
Non-Patent Citations (2)
| Title |
|---|
| "基于FPGA的信号分离器设计";钱苏萍;《中国优秀硕士学位论文全文数据库信息科技辑》;20150415;摘要、正文第1-5章 |
| "基于VoIP技术的VTS通信模拟器设计";耿晓琴;《中国优秀硕士学位论文全文数据库信息科技辑》;20140915;正文第2-3章 |
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