CN105872018A - Medical group communication voice system - Google Patents
Medical group communication voice system Download PDFInfo
- Publication number
- CN105872018A CN105872018A CN201610166220.0A CN201610166220A CN105872018A CN 105872018 A CN105872018 A CN 105872018A CN 201610166220 A CN201610166220 A CN 201610166220A CN 105872018 A CN105872018 A CN 105872018A
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- CN
- China
- Prior art keywords
- audio
- voice
- speex
- opus
- openal
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Classifications
-
- 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
- H04L67/01—Protocols
- H04L67/10—Protocols in which an application is distributed across nodes in the network
- H04L67/104—Peer-to-peer [P2P] networks
- H04L67/1044—Group management mechanisms
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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
- H04L65/40—Support for services or applications
- H04L65/403—Arrangements for multi-party communication, e.g. for conferences
-
- 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]
-
- 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
- H04L65/70—Media network packetisation
-
- 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
- H04L65/75—Media network packet handling
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- Engineering & Computer Science (AREA)
- Multimedia (AREA)
- Computer Networks & Wireless Communication (AREA)
- Signal Processing (AREA)
- Physics & Mathematics (AREA)
- Computing Systems (AREA)
- Mathematical Physics (AREA)
- Theoretical Computer Science (AREA)
- Telephonic Communication Services (AREA)
Abstract
The invention discloses a medical group communication voice system. The medical group communication voice system is successively combined by an audio collection module, an audio processing module, an audio compression module and an audio transmission module; the audio collection module comprises an OpenAL (Open Audio Library); the audio processing module comprises an Speex; the audio compression module comprises an Opus; and the audio transmission module comprises an ORTP. According to the medical group communication voice system provided by the invention, by employing the OpenAl for audio collection, the Speex for audio processing, the Opus for audio encoding and the ORTP for real-time transmission of voice information, and integrating the several modules into one processing procedure, the complete procedure from collecting, processing and compressing to sending of the voice is realized; and meanwhile, multiple people can be supported to communicate in the same voice channel, so the network bandwidth can be reduced, the control of a user side is simplified, and the connectivity of the online user side is guaranteed.
Description
Technical field
The present invention relates to voice technology field, specifically a kind of doctor group leads to voice system.
Background technology
P2P computing (Peer to Peer is called for short P2P) can simply be defined as sharing calculating by directly exchange
Machine resource and service, and the network that P2P computing model application layer is formed is commonly referred to peer-to-peer network.In P2P network environment,
Thousands of the computers being connected to each other are all in reciprocity status, and in general whole network is independent of special concentration clothes
Business device.Each computer in network can serve as the requestor of network service, and sound is made in the request to other computer again
Should, it is provided that resource and service.Generally these resources and service includes: the shared and exchange of information, calculating resource are (such as being total to of CPU
Enjoy), storage share (such as caching and the use of disk space) etc..
Existing voice technology many employings P2P mode, the both sides of call can carry out point-to-point live connection mutually, but
Do not support the most online talk of many people.Do not support particularly with subnetwork the interconnection of P2P mode, even user side all without
Method is carried out.Meanwhile, existing voice technology the most also do not have complete set from voice collecting, process, be compressed to the complete of transmission
Handling process.
Summary of the invention
It is an object of the invention to provide a kind of doctor group supporting that many people converse online and lead to voice system, it is achieved to doctor group
The voice of way system is from the entire flow gathering, processing, be compressed to transmission.
For achieving the above object, the present invention provides following technical scheme:
Doctor group leads to voice system, successively by audio collection module, audio processing modules, audio compression module and audio transmission
Block combiner forms；Described audio collection module includes OpenAL, and described audio processing modules includes Speex, described
Audio compression module includes that Opus, described audio transmission module include ORTP；Described system passes through CopusAl pair
OpenAL, Speex, Opus, ORTP carry out unified integration, it is achieved to the voice of doctor's group's way system from gathering, process, be compressed to
The entire flow sent.
As the further scheme of the present invention: the pre-treatment sequence of described Speex is quiet, the automatic gain of detection, increasing
Benefit, when wherein Speex initializes, frame sign is 10ms frame, and the data of i.e. every 10ms constitute an audio frame.
Compared with prior art, the invention has the beneficial effects as follows:
The doctor group that the present invention provides is led to voice system and is used OpenAL to carry out audio collection, and Speex carries out Audio Processing,
Opus carries out audio coding, ORTP real-time Transmission voice messaging, and by these one handling processes of module integrated one-tenth, it is achieved that
Voice, from gathering, process, be encoded to the entire flow sent, realizes independent of other products.
The present invention provide doctor group lead to voice system use server client pattern, be devoted to solve medical surgery or
Real-time Transmission multi-person speech information during Medical conference, can support that many people converse in same voice channel, converse many people
Time, multi-path voice is done audio mixing, is fused into audio/video flow data with video information, reduce the network bandwidth, simplify user side and control,
The connectedness of the line user end ensured.Doctor's all of voice messaging of group's way system, from transit server, can be located in speech room
Manage many people connection to forward with data, it is achieved many people are online.
Accompanying drawing explanation
Fig. 1 is the pretreatment process figure of the Speex of the present invention；
Fig. 2 is the comparison schematic diagram of Opus audio coding and extended formatting audio coding.
Detailed description of the invention
Below in conjunction with the embodiment of the present invention, the technical scheme in the embodiment of the present invention is clearly and completely described,
Obviously, described embodiment is only a part of embodiment of the present invention rather than whole embodiments.Based in the present invention
Embodiment, the every other embodiment that those of ordinary skill in the art are obtained under not making creative work premise, all
Belong to the scope of protection of the invention.
Embodiment 1
In the embodiment of the present invention, doctor group leads to voice system, as the voice system of doctor's group's way system, successively by audio collection
Module, audio processing modules, audio compression module and audio transmission module combine；Audio collection module includes OpenAL
(Open Audio Library), audio processing modules includes that Speex, audio compression module include Opus, audio transmission module
Including ORTP；Native system passes through the CopusAl unified integration to OpenAL, Speex, Opus, ORTP modules, it is achieved to doctor
The voice of group's way system from gathering, process, be compressed to the entire flow of transmission, the function of concrete each module and realize flow process such as
Under:
One, audio frequency is gathered
Doctor's group's way system uses OpenAL (Open Audio Library) to gather voice data.OpenAL is to provide across flat
The audio API of platform, its design shows to the specially good effect of multichannel three-dimensional position audio.Function main for OpenAL is at source thing
Body, audio buffering and listener encode.Source object comprise one point to the index of relief area, the speed of sound, position and
Direction, and intensity of sound.Listener's object comprises the speed of listener, position and direction, and all the overall of sound increases
Benefit.Comprising the audio data of 8 or 16 bits, monophonic or stereo PCM format in buffering, presentation engine carries out being necessary
Calculate, such as range attenuation, Doppler effect etc..Use OpenAL the most effectively can gather voice letter from sound device
Breath.
The API (Application Programming Interface, application programming interface) that OpenAL provides:
OpenAL gathers voice messaging process
The API explanation of COpusAl
COpusAl is that doctor's group's way system is to Openal, the unified integration of Speex, Opus, oRTP modules.
COpusAL initializes and enters workflow:
Two, Audio Processing
Speex explanation
Speex is a set of audio compression format.Relative to other codec, Speex is well suited for network application, at network
The advantage of oneself uniqueness is had in application.Speex designs in the compress speech of 2-44kbps exclusively for code check, can by 8kHz,
16kHz, 32kHz are compressed in same bit stream, have intensity stereo coding simultaneously, and data-bag lost is hidden, variable bit rate, voice
Catch, discontinuous transmission, fixed-point calculation, the feature of sense organ echo cancellor noise isolation.
The doctor group that the present invention provides is led to voice system and is used Speex to carry out gain, noise reduction, and echo processes, quiet judgement, and
Not responsible coding.
The pre-treatment sequence referring to Fig. 1, Speex is detection quiet (VAD), automatic gain (AGC), gain, wherein
When Speex initializes, frame sign is 10ms frame, and the data of i.e. every 10ms constitute an audio frame.
The API that Speex provides:
The setting macrodeclaration (on (1)/off (2)) of speex_preprocess_ctl
The API explanation of CSpeexPacket:
short data[SIZE_OPUS*2]；
short ref[SIZE_OPUS*2]；
short out[SIZE_OPUS*2]；
if(m_SpeexPacket->DoAEC((short*)data,(short*)ref,(short*)out)){
/*
Out is exactly the result after processing, and the size of data, ref, out must be consistent
*/
memcpy(ref,out,sizeof(out))；
}else{
memcpy(ref,out,sizeof(data))；
}
Three, audio compression
Opus illustrates:
Doctor group is led to voice system and is used Opus to encode audio frequency.Opus encoder is the lattice damaging acoustic coding
Formula, is come in develop by Internet Engineering Task group (IETF), it is adaptable to the live sound transmission on network, it can be low bit-rate
Arrowband becomes high-quality three-dimensional voice.It support 6kb/s-510kb/s bitrate range, sample rate from 8kHz to 48kHz,
Frame sign, from 2.5ms to 60ms, is supported monophonic and stereo, can dynamically be regulated bit rate, amount of bandwidth and frame sign.
The predecessor of Opus is celt encoder.In current speech coded format, the performance of Opus is distinguished.With AAC form
Comparing, by many contrast tests, under low bit-rate, Opus defeats HE AAC the most with the obvious advantage completely, and middle code check the most may be used
The AAC form of about 30% is exceeded with the enemy's code check that is equal to, and closer to original audio, concrete audio coding result ratio under high code check
More as shown in Figure 2.
The API that Opus provides:
Corresponding grand of opus_encoder_ctl attribute
The API explanation of COpusPacket:
Four, audio frequency sends
ORTP illustrates:
Doctor group is led to voice system and is used ORTP to carry out the transmission of voice messaging.
RTP (Real-timeTranspoRTProtocol) is the one on Internet for multimedia data stream
Host-host protocol, does the application in terms of streaming media and be unable to do without realization and the use of Real-time Transport Protocol, in order to more rapidly in project
Middle application Real-time Transport Protocol realizes the transmission of Streaming Media, and we typically can select to use some RTP storehouses, and ORTP is to achieve Real-time Transport Protocol
One increase income storehouse, this storehouse pure use c language is write.
The API that ORTP provides:
The api interface of CORTPSender:
Doctor group leads to defmacro:
The doctor group that the present invention provides is led to voice system and is used OpenAL to carry out audio collection, and Speex carries out Audio Processing,
Opus carries out audio coding, ORTP real-time Transmission voice messaging, and by these one handling processes of module integrated one-tenth, it is achieved that
Voice, from gathering, process, be encoded to the entire flow sent, realizes independent of other products.
The present invention provide doctor group lead to voice system use server client pattern, be devoted to solve medical surgery or
Real-time Transmission multi-person speech information during Medical conference, can support that many people converse in same voice channel, converse many people
Time, multi-path voice is done audio mixing, is fused into audio/video flow data with video information, reduce the network bandwidth, simplify user side and control,
The connectedness of the line user end ensured.Doctor's all of voice messaging of group's way system, from transit server, can be located in speech room
Manage many people connection to forward with data, it is achieved many people are online.
It is obvious to a person skilled in the art that the invention is not restricted to the details of above-mentioned one exemplary embodiment, Er Qie
In the case of the spirit or essential attributes of the present invention, it is possible to realize the present invention in other specific forms.Therefore, no matter
From the point of view of which point, all should regard embodiment as exemplary, and be nonrestrictive, the scope of the present invention is by appended power
Profit requires rather than described above limits, it is intended that all by fall in the implication of equivalency and scope of claim
Change is included in the present invention.
Although moreover, it will be appreciated that this specification is been described by according to embodiment, but the most each embodiment only wraps
Containing an independent technical scheme, this narrating mode of description is only that for clarity sake those skilled in the art should
Description can also be formed those skilled in the art through appropriately combined as an entirety, the technical scheme in each embodiment
May be appreciated other embodiments.
Claims (2)
1. doctor group leads to voice system, successively by audio collection module, audio processing modules, audio compression module and audio transmission mould
Block combines；It is characterized in that, described audio collection module includes that OpenAL, described audio processing modules include
Speex, described audio compression module includes that Opus, described audio transmission module include ORTP；Described system is passed through
CopusAl carries out unified integration to OpenAL, Speex, Opus, ORTP, it is achieved to doctor group's way system voice from gather, from
Manage, be compressed to the entire flow of transmission.
Doctor group the most according to claim 1 leads to voice system, it is characterised in that the pre-treatment sequence of described Speex is
Detecting quiet, automatic gain, gain, when wherein Speex initializes, frame sign is 10ms frame, and the data of i.e. every 10ms constitute one
Audio frame.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201610166220.0A CN105872018A (en) | 2016-03-21 | 2016-03-21 | Medical group communication voice system |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201610166220.0A CN105872018A (en) | 2016-03-21 | 2016-03-21 | Medical group communication voice system |
Publications (1)
| Publication Number | Publication Date |
|---|---|
| CN105872018A true CN105872018A (en) | 2016-08-17 |
Family
ID=56625393
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CN201610166220.0A Pending CN105872018A (en) | 2016-03-21 | 2016-03-21 | Medical group communication voice system |
Country Status (1)
| Country | Link |
|---|---|
| CN (1) | CN105872018A (en) |
Citations (2)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN103888712A (en) * | 2014-01-28 | 2014-06-25 | 中译语通科技（北京）有限公司 | Multilingual synchronous audio and video conference system |
| CN204291130U (en) * | 2014-12-19 | 2015-04-22 | 罗晓东 | Mobile network is utilized to realize equipment that is single and multi-person speech message |
-
2016
- 2016-03-21 CN CN201610166220.0A patent/CN105872018A/en active Pending
Patent Citations (2)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN103888712A (en) * | 2014-01-28 | 2014-06-25 | 中译语通科技（北京）有限公司 | Multilingual synchronous audio and video conference system |
| CN204291130U (en) * | 2014-12-19 | 2015-04-22 | 罗晓东 | Mobile network is utilized to realize equipment that is single and multi-person speech message |
Non-Patent Citations (2)
| Title |
|---|
| 谢晓钢，蔡骏，陈奇川，欧建林: ""基于Speex语音引擎的VoIP系统设计与实现"", 《计算机应用研究》 * |
| 陈旭升: ""基于PJSIP的音视频通信及录放系统研究与实现"", 《中国优秀硕士学位论文全文数据库信息科技辑》 * |
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Application publication date: 20160817 |