2002 Vol. 27, No. 1
In this paper the authors first analyze the requirement and feasibility of the integration of spatial information and mobile communications,then discuss the key technologies of the system integration include the technology of wireless transmission and wireless interlink,the technology of spatial data management,the method of spatial data description and presentation,the client-side programming technology,and the position determining equipment.A three layers architecture is brought forward to integrate the spatial data management and mobile device.Finally,we summarize the application field and market foreground according to the needs of personal,enterprise an d government and enumerate the typical application of the integrated system such as the express logistics system,public information service,intelligent transportation system etc.
The system structure,subsystem and core techniques including video coding rate control,video signal pre and post processing,quick algorithm for heavy density calculation module,lip synchronization and audio-video balance control under bad communication condition,are talked ab out in the paper.The system has the features such as crisp and smooth video performance,fr am e refresh rate of up to 25 f/s,less than 300ms system one-way processing delay,proprietary technology to support steady video communications at bad line condition down to 12kbit/s,et al.
This paper introduces the basic principle of PSTN video-phone and V.80 protocol.By analyzing the Recommendation V.80 which is in-band DCE control and synchronous data modes for asynchronous DTE,the author implemented an effective control model of synchronous trans-mission for the characteristics of low bit-rate multimedia communication on PSTN.It improved on the performance of H.324 videophone(VP324) greatly,Based the control mechanics,the video phone system can interact with other products based H.324 effectively.The increasing performances of the VP324 system come from the following aspects:① Adjusting the video codec strategies dynamically to adapt to current real communicating bandwith.It can not only use the bandwith sufficiently,but also prevent data overflow of multiplexing layer,and reduce the delay time of media signal;② Putting forward an Autobalancing Technology for audio-video processing which makes the picture more smooth and clear.It vacate more bandwith spaces for video stream by transferring the audiocodec policies which reduce the occupancy of audio streaming bandwith between 5.3k/6.3k and 2.4k;③ Reducing the effect on lip-synchronization to the lowest level due to bandwith waving dynamically,Through the V.80 control module cooperating with H.223 module and media codec modules.VP324 system already passed the professional accreditation of National Information Industrial Bureau,its all performance indexes achieve the national accredited requirements,and some of them exceed the other products based H.324,such as the highest frame rate is up to 25 fps under PSTN 28.8k bandwith.
This paper describes the complete procedure of framework design on a video coding system for error-prone heterogenous network environment.First of all,the requirements of the system are analyzed and the targets of the system design are emphasized on three categories:①improving video performance under the constraints of network bandwidth and computational complexity,②providing error robust mechanism when packet loss occur,③adding capability of layered video coding for heterogenous network transmission.Then,ITU-T H.263+recommendation is introduced,especially about its 16 optional enhanced coding modes:feature,effect and benefit.The system design on mode selection is somewhat a trade-off.Five enhanced coding modes are preferred for our purpose:advanced INTRA coding mode,deblocking filter mode,modified quantization mode,slice structured mode and temporal,SNR and spatial scalability mode.At last,some useful and important system elements beyond H.263+recommendation are discussed:RTP packetization,error concealment and error tracking.Application shows that all the targets are easily touched by the implementation based on this design.
In this paper a brief introduction to the fundamental principle of ITU-T G.723.1 encoding decoding algorithms is firstly carried out.Subsequently,some main features about the fixed-point DSP chip TM1300 such as hardware structure,the principles of audio input and output,embedded operating system running on TM1300 DSPCPU are introduced.On the basis of the above introduction,this paper designs a set of feasible project running on TM1300 for real time speech sampling,speech encoding,decoding and playing aim to hardware structure characteristic of the chip.In order to come up to the real time requirement,the fixed-point algorithm of G.723.1 is improved with this understanding of passing all test sequences provided by ITU-T.The key technologies with real time implementation of G.723.1 speech coder on TM1300 is discussed emphatically in this paper;which mostly include the design with systemic overall framework,material algorithm optimizations and the principle of program code optimizations which is generalized according to the structural feature of TriMedia DSPCPU.At the same time,some problems which are met in the process of specific realization are also discussed.In addition,the corresponding resolution are brought forward in this paper.The experiment shows that at least 5 ways of speech encoder and decoder can be carried out on single TriMedia 1300 simultaneously.
As BCH is a kind of linear error-correcting code with good performance up to now,it has been used widely.But for the relatively longer codes,it cost large computation,so special hardware is alwayes employed in real-time communication.But in low-cost communication system or desktop communication system with no hardware support,the most urgent problem is how to complete the computation of BCH code efficiently,namely the algorithm efficiency of BCH code.In this paper,firstly,coding method and error-correction theory of BCH(511,493) error-correcting in video coding recommendation H.261 are introduced,then the low efficiency in classical division algorithm with software completion is analyzed.Subsequently,a kind of quick algorithm on correcting 2-bit random errors is offered.With practical evidence,this algorithm can increase the whole running speed at least 20 times than the classical division algorithm and meet the requirement in real-time video communication.It has been successfully used in PSTN and ISDN videophone.
ISDN can supplied a point-to-point digital connection that can be used to carry multi-traffic including both audio and non-audio signals and perform many operations such as dialing to access the Internet,and video phone especially.In this paper,the special driver model for a video phone ISDN card,which hardware was designed for multiplexed multifunction,to access the Internet is developed.The related ISDN network communication protocols are discussed in detail and the principle of the ISDN multimedia terminal card accessing the Internet under the scheme of MS-Windows RAS(remote access system) is analyzed.In the structure of RAS,the call control in the line is managed by TAPI(Telephone API),and the data transport is achieved by transport drivers.NIC(Network interface card) drivers must complete four functions:to complete the protocols Q.931 and Q.921 in ISDN D channel and control the communication in physical channels;to receive or send data in B channel;to function the interface with TAPI and the management of link layer;to act as the interface with transport drivers by WAN function library and NDIS function library.A miniport driver based on NDIS(network driver interface specification) for the ISDN multimedia card,which enables the card to function as an ISDN modem and a multimedia terminal simultaneously,is developed and the design guide and implementation details are reported.
In order to realize the computer screen share in distant teaching,distant stocking and distant cooperation working,a compression scheme is proposed for graphiclike image sequences(GISS) in this paper.It first converts the graphiclike image to index image,then codes them with interframe run length algorithm and adaptive predictive Huffman algorithm,finally we code stream with LZ77 algorithm.The scheme employs the intrinsic characters of the GISS——the color distribution of the image mainly concentrates on few kinds of color and the pixels which have the same color are also adjacent in space.The result of computer simulation shows that our sheme can compress the GISS effectively and improves the compression ratio obviously relative to the traditional lossless compression algorithms.The most compression ratio is 80:1.
The software layer that describes the interface between the software on chip and any peripherals on the board is known as the board support package,or BSP.BSP allows to change the boards design without affecting the software that has already been developed.This paper tries to describe a method of communication between application program and BSP and organizing of functions in BSP which work with Philips TM1x00 CPU and pSOSystem embedded operating system,or pSOS.TM1x00 is a media processor for high-performance multimedia applications that deal with high-quality video and audio.pSOS is a modular,high-performance real-time operating system designed specifically for embedded microprocessors.A basic BSP consists of many files about every peripheral.One of those has a few functions which initialize corresponding peripheral and finish the function that application program needs.This paper brings forward a method to connect the application program and functions in the BSP.The method adds a series of libraries which make users application program need not consider the hardware.It can only call the library to finish the operation about hardware.Accordingly the library calls the functions in BSP to control hardware or get information about hardware and return to application.In addition,this paper uses a chain table to manage the functions in BSP.These functions'name are defined as functions'pointer.These pointer are put into a structure.Functions'pointer variables about the same peripheral are all put into a structure.The address of the structure is added to chain table.So all peripheral structure form a chain table.Library searches the chain table to obtain the address of structure and call the corresponding function.Of course,the chain table should be built during the system startup and before the application program of user is executed.Then when application program is running,it can call the library to use chain table in any time.This article provides a thought of realizing the BSP communicating with application program.The application program of user is very clear when the librares are used to be an interface between it and BSP.Use of the chain table makes it easy to manage the BSP.When hardware needs to be upgraded,the chain table makes the change for BSP appear to be in proper order.So these two methods have some value in realizing it.
Using the principle of oscillograph on the analysis of hardware,we introduce the assistant application of real-time monitor observer in the development of audiovisual communication system.In this paper,we emphasize on introducing it's use in the development of videophone based on ISDN H.320 series protocols.Using the observer,we define kinds of"probes" independent of real communication system.Using these probes to monitor running system,dynamical parameters are snatched.Base on these parameters,the system characteristics are realized.Thus we can find the system's"bottle-neck" as soon as possible and adjust system design and optimize system.In the development of video module,we define parts of on-line monitoring windows,from which we can get statistic data;with which we adjust arithmetic in video coding module and decoding module.The same application in video module can help to control coding efficiency in video coding module,and help us to choose accurate video format in communication.Software probes in the application of the buffer in thread help us to know whether the buffer size is suitable,and make policy that we read and write data to those buffers.For example,according to on-line observing,we finally choose quantity and size of the audio buffer to achieve the best lip-synchronization result.
In electrical power industry,optical sensors are inherently advantageous over conventional measurement techniques.In this paper,the research and developments of some high voltage optical current and optical voltage transducers are reported in detail.A three-phase optical current transducer was developed and put into use in a real 110kV electrical substation.The calibrated ratio errors within rated current range is less than 0.2%.By employing optical insulators,the optical current transducers are with good insulation,light weight and small size.At the same time,the combined current and voltage optical metering unit(OMU) for 500kV power system is presented.In the OMU,the BGO cystal and flint glass were used as sensing elements for high voltage and electrical current,respectively.The experimental results showed that both optical current sensor and optical voltage sensor were of good linearity.The specifically designed insulator for 500kV high voltage system was a optical insulator embedded in a conventional porcelain column.Some new technologies used in optical current sensors recently are discussed as well.These new sensors include:bulk glass optical current sensor with an adjustable multi-ring closed optical path,hybrid type bulk glass sensor with multiple critical angle reflections for sensitivity enhancement,right-triangle shaped detachable Faraday effect current clamp and fiber Bragg grating based magnetostrictive sensor for DC current measurements.The optical fiber technology used in power system in the future for optical sensing,optical monitoring,optical communication,optical relay protection and optical local area network are introduced as a conclusion.
The ITU-T recommendation H.263 specifies a coded representation that can be used for compressing the moving picture component of audio-visual services at low bit rates.The basic configuration of the video source coding algorithm is based on Recommendation H.261 and is a hybrid of inter-picture prediction to utilize temporal redundancy and transform coding of the remaining signal to reduce spatial redundancy.TM1300 is a media processor for high-performance multimedia applications that deals with high-quality video and audio.TriMedia's DSPCPU family delivers exceptional performance and high-level language programmability for multimedia applications through the use of its VLIW architecture.TriMedia's VLIW architecture combines innovations in compiler and software design with advances in logic design.pSOSystem is a modular,high-performance real-time operating system designed specifically for embedded microprocessors.It provides a complete multitasking environment based on open systems standards.The feasible video codec running on TM1300 is designed according to the characteristic of the hardware structure.It includes real-time video signal sampling,video codec,video playing and so on.The emphatically key technologies and problem of the real-time implementation of H.263 video codec on TM1300 are discussed.The optimization technical is given in detail to the H.263 encoding and decoding algorithms for this hardware and compile system.This paper also describes several optimization ways,such as using custom operation,loop unrolling,disposing variable and function,decision tree grafting,compile and algorithm optimization,and various optimization methods supported by the TriMedia compilation system as well as techniques for exploiting the fine-grain parallelism of the TriMedia architecture.The test results are listed in the end.
he market of surveillance system is expanding rapidly.The very surveillance product,which is according with the international communications standards and easy to be customized,is most welcome to customers and applied widely in all trades.On the other hand,those nonstandard products,only be used in local area,are gradually washed out.This digital remote surveillance system based on ITU-T recommendation can be used in PSTN/ISDN/LAN for remote surveillance except in the local area.The core technology of the system lies in the multimedia application SDK(software development kit) based on ITU-T H.324,ITU-T H.323,ITU-T H.320.It encapsulates the abilities of video/audio/data processing and the complexity of the protocol.It comprises eight kinds of API service functions and messages for the application of surveillance system.The multimedia SDK adapts the surveillance system to different network environments in a simple and effective way.At last this paper describes its application in the electric equipment remote surveillance system.
MCU(Multi-point Control Unit) is the key equipment of multi-point video conference.Multi layer network connected with multi MCUs can implement large-scale video conference.Because of their characteristic of unbounded geographical zone,multi MCUs are set on each floor in a building or at another office building,or different zones.It brings about the problem how to manage and control dispersing MCUs.Creating system of controlling MCUs in long-distance is the optimal scheme to solve this problem.This paper analyzes not only the characteristic of multi-layer distributed system structure provided by the system of controlling MCUs in long-distance,but also the special requests in application of the Internet.The methods to select thread type of component,design interface,arrange component physical location and test component in idiographic application condition are introduced.And how to unite COM and ASP to create COM components in ASP applications is discusseed in detail.Using the technology in the paper,programmer can set up greater Internet applications.They have good performance and multi-user.At last,how to select COM's security and its lifecycle in COM's design and realization is discussed.
In the block-based transform image coding techniques,especially in the video coding at very low bit rate,block artifacts influence the subjective quality seriously.In this paper we propose an adaptive post-processing algorithm to reduce the block artifacts,which has the property of low computational burden and obvious effect.Simulation results show that the proposed algorithm can improve the subjective quality and objective PSNR of the coded image.
Owing to the flexibility of time-frequency resolution of wavelet transform,we present an improved voice activity detector(VAD) based on wavelet transform.Robust parameters in different scale and time resolution are computed for VAD decision,such as silence measure,stability measure of amplitude spectrum between adjacent frames,background noise measure of different frequency band,time-domain stability measure of scale 1.The silence measure is used to detect the existence of silence in the input frame.The stability measure of amplitude spectrum between adjacent frames is adopted to give a rough decision of the detection of stable noise based on the assumption that background noise is stable.If current input frame is noise,the energy of every frequency band is below the average of background noise energy threshold over long time.We divide the signal bandwidth into several scales by wavelet transform,and calculate the background noise measure of different scale.In low scale the input signal changes rapidly,and the variety of short time energy will be removed with long window.We calculate the mean square error of short time energy,and get time-domain stability measure from detail coefficients of scale 1.With these measures,we make the VAD decision.Compared with G.729 Annex B,the authors can detect the voice activity more accurately and reduce the ratio of speech clipping using the new algorithm.And the improved algorithm can achieve robust performance for different background noise,even in serious low signal-to-noise environment about 10dB.
This paper analyses the future development of the standalone IP videophone and its excellence in comparison with tradition videophone,expatiates on the application of the standalone IP videophone and introduces the structure of H.323 system and some protocols which include video code,audio code,system control(H.245 control,H.225 call control and RAS control) and H.225 layer.This paper puts forward a project of standalone IP videophone with the frame of H.323 protocol which includes two parts,hardware and software.The hardware part mostly consists of CUP processor,video processor,audio processor.CPU processor is made up of RA-RISC,ROM and SDRAM and employs W90221 chip,W9961 chip and W9975 chip.W90221 is connected with network processor and video processor by PCI bus and with audio processor by synchronization series interface(SSI).W9961 chip compresses and decompresses video signals,which is based on standard ITU-T H.261 and H.263 protocols and can work with 32-bit RISC Processor.W9975 chip can apply the videophone with H.323 or H.324 system and IP videophone with IP network.The software part mostly includes drivers and applications.The drivers are used to manage the hardware resources and serve the applications as library.the drivers are made up of audio-base,Video-base,interface-base,TCP/IP protocol-stack and H.323 protocol-stack.The applications include seven tasks:H.323 main task,net task,user input task,monitor task,message dispatcher task,G723 transmit task and H.263 transmit task.They achieve the operations of communication,synchronization,mutex through message queens,event,signal lamp,and finish the system functions unanimously.The project imports embedded operation system Super-Task to manage and control the system resources.
Most of the mandarin text-to-speech systems are syllable-based which only include syllable-internal coarticulation while cross out any cross-syllable coarticulation.One solution to this problem is to abandon syllable-based models in favor of units which can model both syllable-internal and cross-syllable coarticulation.One such unit is the diphone which has been quite used in English TTS sysem.The concept of diphone can be improved in Chinese speech synthesis for there are only 410 syllables in the mandarin.It is shown that diphone-based models can model cross-syllable coarticulation and produce natural-sounding speech rather than syllable-based systems which just insert silence between syllables.
ITU-T H.323 describes the components for multimedia communication systems in those situations where the underlying transport is a packet-based network.The multipoint control unit(MCU) can provide centralized processing of audio,video,and/or data stream in a multipoint conference.MCU is composed of the multipoint processor(MP) and the multipoint controller(MC).MP takes responsibilities of collecting audio,video,and/or data streams from all the terminals of the multipoint conference,processing all the information in the streams,and sending the processed data to the appointed terminals under the control of MC.In this paper,the authors bring forward some solutions for the request of processing audio stream,and then particularly present a practical policy aiming at the audio signals mixing operation.In the centralized multipoint conference mode,it is necessary to do the audio mixing operation on the speech from all the audio channels.The basic audio mixing technology includes three steps.First,MCU decodes the audio code streams from every audio channel respectively,and gets the sum of all the decoded speech.Second,the target speech corresponding to every terminal is gained after subtracting the source signal from the sum.Lastly,the target speech of every terminal is coded respectively,and transmitted to the specific terminal.So each of the terminals receives the audio signal containing all the signal of other terminals.There are many shortcomings in the method above.First,the more the terminals accessing the videoconference are,the more number of speech Codec used by MCU consequently is.Thus the calculating burden of MCU becomes heavy.Second,it is not necessary to mix all the speech from every audio channel equally.It is difficult for the perceptual ability to distinguish the useful information when the speech signals taken into the audio-mixer are more than 4 channels.Therefore,we design an improved audio-mixer that employs a kind of competitive mechanism.When the number of terminals accessing MCU is more than 4,we select 4 channels with the higher speech energy within fixed time interval and take them into the audio-mixer.The speech signals of other channels are regarded as the background noise after a certain of attenuation.The audio-mixer calculates the energy of speech in a fixed time interval and decides the state of every channel according to their speech energy.The state of every channel is preserved until to the end of the following time interval.
This paper describes a new IP transport protocol,the stream control transmission protocol(SCTP),which is at an equivalent layer as UDP(User Datagram Protocol) and TCP(Transmission Control Protocol) and especially fit for signaling messages transmission such as multimedia data over IP networks.The infrastructure of the SCTP which enables a reliable transport service through an connection-oriented mechanism is introduced,and the core features of the SCTP,multi-streaming and multi-homing are discussed.In addition,the SCTP message format,state machine and the three main phases in states transferring,and the specific mechanisms to obtain appropriate congestion avoidance and resistance to flooding and masquerade attacks during these phases are analyzed.Finally,the details of the implementation of SCTP are presented.