Fundamental and Technological Limitations of Immersive Audio Systems
CHRIS KYRIAKAKIS,MEMBER,IEEE
Numerous applications are currently envisioned for immersive audio systems.The principal function of such systems is to synthe-size,manipulate,and render sound?elds in real time.In this paper, we examine several fundamental and technological limitations that impede the development of seamless immersive audio sys-tems.Such limitations stem from signal-processing requirements, acoustical considerations,human listening characteristics,and listener movement.We present a brief historical overview to outline the development of immersive audio technologies and discuss the performance and future research directions of immersive audio systems with respect to such https://www.doczj.com/doc/6918137144.html,st,we present a novel desktop audio system with integrated listener-tracking capability that circumvents several of the technological limitations faced by today’s digital audio workstations.
Keywords—Acoustic signal processing,audio systems,auditory system,multimedia systems,signal processing.
I.I NTRODUCTION
Emerging integrated media systems seamlessly combine digital video,digital audio,computer animation,text,and graphics into common displays that allow for mixed media creation,dissemination,and interactive access in real time. Immersive audio and video environments based on such systems can be envisioned for applications that include teleconferencing and telepresence;augmented and virtual reality for manufacturing and entertainment;air-traf?c con-trol,pilot warning,and guidance systems;displays for the visually or aurally impaired;home entertainment;distance learning;and professional sound and picture editing for television and?lm.The principal function of immersive systems is to synthesize multimodal perceptions that do not exist in the current physical environment,thus immersing users in a seamless blend of visual and aural information. Signi?cant resources have been allocated over the past20 years to promote research in the area of image and video processing,resulting in important advances in these?elds.
Manuscript received September8,1997;revised December4,1997.The Guest Editor coordinating the review of this paper and approving it for publication was T.Chen.
The author is with the Integrated Media Systems Center,University of Southern California,Los Angeles,CA90089-2564USA(e-mail: ckyriak@https://www.doczj.com/doc/6918137144.html,).
Publisher Item Identi?er S0018-9219(98)03283-6.On the other hand,audio signal processing,and particularly immersive audio,have been largely neglected.
Accurate spatial reproduction of sound can signi?cantly enhance the visualization of three-dimensional(3-D)infor-mation for applications in which it is important to achieve sound localization relative to visual images.The human ear–brain interface is uniquely capable of localizing and identifying sounds in a3-D environment with remarkable accuracy.For example,human listeners can detect time-of-arrival differences of about
7
such fundamental limits,and delineation of technological considerations that affect present and future system design and development.In the?nal sections,we will present a novel sound-reproduction system that addresses several of the current technological limitations that currently affect the quality of audio at the desktop.This system incorporates a video-based tracking method that allows real-time process-ing of the audio signal in response to listener movement. II.T HE N ATURE OF L IMITATIONS IN
I MMERSIVE A UDIO S YSTEMS
There are two classes of limitations that impede the implementation of immersive audio systems.The?rst class encompasses fundamental limitations that arise from phys-ical laws,and its understanding is essential for determining the feasibility of a particular technology with respect to the absolute physical limits.Many such fundamental limitations are not directly dependent on the choice of systems but instead pertain to the actual process of sound propagation and attenuation in irregularly shaped rooms.The physical properties of the acoustic environment are encoded in the sound?eld and must be decoded by an immersive audio system in order to accurately simulate the original envi-ronment.The in?uence of the local acoustic environment is re?ected in the perception of spatial attributes such as direction and distance,as well as in the perception of room spaciousness and source size[1],[2].The situation is further complicated by the fact that the decoding process must include the transformations associated with human hearing.These include the conversion of spatial sound cues into level and time differences and direction-dependent frequency-response effects caused by the pinna,head,and torso through a set of amplitude and phase transformations known as the HRTF’s.The seamless incorporation of such cues in immersive audio systems is a very active area of research that,if successful,will give rise to systems that begin to approach performance near the fundamental limits. The second class of limitations consists of constraints that arise purely from technological considerations.These are equally useful in understanding the potential applications of a given system and are imposed by the particular tech-nology chosen for system implementation.For example, the process of encoding parameters associated with room acoustics into sound?elds can be modeled using numerical methods.In theory,this would involve the solution of the wave equation for sound subject to the boundary conditions dictated by the complex(absorptive,re?ective, and diffractive)room surfaces.The computational com-plexity of this problem is very high and involves the calculation of estimated10normal modes that fall within the range of human hearing(20Hz–20kHz)for a large hall[4].More recent methods have been developed for rendering sound?elds through a process called auralization. Such methods utilize a combination of scaled models, digital?ltering,and special-purpose hardware for real-time convolution to predict and render the sound?eld[5].As the processing power of digital signal processing(DSP)hardware increases,the capability of auralization systems to render complex sound?elds will increase proportionally. III.B RIEF H ISTORICAL O VERVIEW
A.Two-Channel Stereo
Although many of the principles of stereophonic sound were developed through research efforts in the early1930’s, there still remains a misconception as to the meaning of the word“stereo”itself.While it is generally associated with sound reproduction from two loudspeakers,the word originates from the Greek stereos,meaning solid or three-dimensional.The two-channel association came about in the1950’s because of technological limitations imposed by the phonograph record that had only two groove walls for encoding information.
Stereophony started with the work of Blumlein[6]in the United Kingdom,who recognized early on that it was possible to locate a sound within a range of azimuth angles by using an appropriate combination of delay and level differences.His work focused on accurate reproduction of the sound?eld at each ear of the listener and on the de-velopment of microphone techniques that would allow the recording of the amplitude and phase differences necessary for stereo reproduction.Fletcher,Steinberg,and Snow at Bell Laboratories in the United States[7]–[9]took a differ-ent approach.They considered a“wall of sound”in which an in?nite number of microphones is used to reproduce a sound?eld through an in?nite number of loudspeakers, similar to the Huygens principle of secondary wavelets. While this made for an interesting theoretical result,the Bell Labs researchers realized that practical implementations would require a signi?cantly smaller number of channels. They showed that a three-channel system consisting of left,center,and right channels in the azimuth plane could represent the lateralization and depth of the desired sound ?eld with acceptable accuracy[Fig.1(a)].The?rst such stereophonic three-channel system was demonstrated in 1934with the Philadelphia Orchestra performing remotely for an audience in Washington,DC,over wide-band tele-phone lines.
B.Four-Channel Matrixed Quadraphonic System
While stereophonic methods can be a powerful tool in the reproduction of the spatial attributes of a sound ?eld,they fall short of true three-dimensional reproduction. The quadraphonic system attempted to circumvent such limitations by capturing and transmitting information about the direct sound and the reverberant sound?eld[10],[11]. To deliver the four channels required by quadraphonic recordings over a two-channel medium(e.g.,the phono-graph record),it was necessary to develop an appropriate encoding and decoding scheme.Several such schemes were proposed based on4:2:4matrix encoding/decoding that relied on phase manipulation of the original stereo signals [12].Quadraphonic systems were capable of reproducing sound images fairly accurately in the front and rear sectors
942PROCEEDINGS OF THE IEEE,VOL.86,NO.5,MAY1998
(a)(b)
Fig.1.(a)Stereo was originally invented based on three loud-speakers.Sound images in the center are rendered by a real loudspeaker that delivers the same direct sound to both ears.(b)In
the two-loudspeaker stereo con?guration with a phantom center,
the cross-talk terms give rise to a less stable center image as well
as a loss of clarity.
of the azimuthal plane,but they exhibited serious limita-tions when attempting to reproduce sound images to the side of the listener.Experiments showed[13],[14]that this was a limitation associated as much with sound-?eld synthesis using only four channels as with human psychoacoustic mechanisms.These technical limitations and the presence of two competing formats in the consumer marketplace contributed to the early demise of quadraphonic systems.
C.Multichannel Surround Sound
In the early1950’s,the?rst multichannel sound format was developed by20th Century Fox.The combination of wide-screen formats such as CinemaScope(35mm) and six-track Todd-AO(70mm)with multichannel sound was the?lm industry’s response to the growing threat of television.Stereophonic?lm sound was typically repro-duced over three front loudspeakers,but these new formats included an additional monophonic channel that was repro-duced over two loudspeakers behind the audience and was known as the effects channel.This channel increased the sense of space for the audience,but it also suffered from a serious technological limitation.Listeners seated on-center with respect to the rear loudspeakers perceived“inside-the-head”localization similar to the effect of stereo images reproduced over headphones.Listeners seated off-center localized the channel to the effects loudspeaker that was closest to them as dictated by the law of the?rst-arriving wavefront,thus destroying the sense of envelopment de-sired[14][Fig.2(a)].The solution to these problems was found by introducing a second channel reproduced over an array of loudspeakers along the sides of the theater to create a more diffuse sound?eld[Fig.2(b)].
In the mid-1970’s,a new sound technology was in-troduced by Dolby Laboratories called Dolby Stereo.It was based on the optical technology that had been used for sound on?lm since the1930’s,and it circumvented the problems associated with magnetic multitrack record-ing.Dolby developed a matrix method for encoding four channels(left,center,right,and mono surround)into two channels using a technique derived from the matrix
methods
(a)(b)
Fig.2.(a)In early surround-sound systems with a mono surround channel,listeners seated off-center perceived the sound as if it originated from the effects loudspeaker that was closest to them,
thus destroying the desired sense of envelopment.(b)Current sys-
tems use stereo surrounds reproduced over an array of loudspeakers along the sides of the theater to create a more diffuse sound
?eld.
Fig.3.Current commercial multichannel systems encode the LFE,three front,and two surround channels into a bit stream that
is decoded at the user’s end.With proper loudspeaker selection
and placement,it is possible to simulate the experience of a movie theater.Dipole surround loudspeakers that do not radiate sound directly in the direction of the listener’s ears produce the best envelopment.
used in quadraphonic systems but also ensured mono and stereo backward compatibility.In1992,further enhance-ments by Dolby were introduced through a new format called Dolby Stereo Digital(SR?D).This format eliminated matrix-based encoding and decoding and provided?ve discrete channels(left,center,right,and independent left and right surround)in a con?guration known as stereo sur-round.A sixth,low-frequency-enhancement(LFE)channel was introduced to add more head room and prevent the main speakers from overloading at low frequencies.The bandwidth of the LFE channel is limited between0and120 Hz,a frequency regime that is outside the localization range for human listeners in a reverberant room,thus simplifying the placement requirements for the subwoofer used for LFE reproduction(Fig.3).
Recent advances in digital audio compression and optical storage have made it possible to deliver up to six discrete audio channels in a consumer format centered around
KYRIAKAKIS:LIMITATIONS OF IMMERSIVE AUDIO SYSTEMS943
the Dolby AC-3compression scheme.1With exciting new formats such as an audio-only digital video disc just around the corner,the number of channels could easily increase to ten or more.While there are several million consumer systems capable of reproducing more than two channels, the majority of users(particularly those with desktop com-puter systems)would?nd the use of multiple loudspeakers impractical.In the sections that follow,we examine the requirements of systems that allow delivery of multiple channels over two loudspeakers using DSP to simulate certain characteristics of human listening.
IV.S PATIAL(3-D)A UDIO
A.Physiological Signal Processing
The human hearing process is based on the analysis of input signals to the two ears for differences in intensity, time of arrival,and directional?ltering by the outer ear. Several theories were proposed as early as1882[15]that identi?ed two basic mechanisms as being responsible for source localization:1)interaural time differences(ITD’s) and2)interaural level differences(ILD’s).A later theory by Lord Rayleigh[16]was based on a combination of ITD and ILD cues that operated in different wavelength regimes. For short wavelengths(corresponding to frequencies in the range of about4–20kHz),the listener’s head casts an acoustical shadow giving rise to a lower sound level at the ear farthest from the sound source(ILD)[Fig.4(b)].At long wavelengths(corresponding to frequencies in the range of about20Hz–1kHz),the head is very small compared to the wavelength,and localization is based on perceived differences in the time of arrival of sound at the two ears (ITD)[Fig.4(a)].The two mechanisms of interaural time and level differences formed the basis of what became known as the duplex theory of sound localization.In the frequency range between approximately1and4kHz,both of these mechanisms are active,which results in several con?icting cues that tend to cause localization errors. While time or intensity differences provide source di-rection information in the horizontal(azimuthal)plane, in the median plane,time differences are constant and localization is based on spectral?ltering.The re?ection and diffraction of sound waves from the head,torso,shoulders, and pinnae,combined with resonances caused by the ear canal,form the physical basis for the HRTF.The outer ear can be modeled(in the static case)as a linear time-invariant system that is fully characterized by the HRTF in the frequency domain.As Blauert[17]describes it,the role of the outer ear is to superimpose angle-and distance-speci?c linear distortions on the incident sound signal.Spatial information is thus encoded onto the signals received by the eardrums through a combination of direction-dependent and direction-independent?lters[18],[19].The magnitude and phase of these head-related transfer functions vary signi?cantly for each sound direction but also from person to person.
1See Dolby Laboratories at
https://www.doczj.com/doc/6918137144.html,.
(a)(b)
Fig.4.(a)In the low-frequency regime,sound is localized based
on differences in the time of arrival at each ear.(b)At higher frequencies,the wavelength of sound is short relative to the size
of the head,and localization is based on perceived level differences caused by head shadowing.In the intermediate-frequency region,
both mechanisms are active,and this can give rise to con?icting cues.
The emerging?eld of3-D audio is based on digital implementations of such HRTF’s.In principle,it is possi-ble to achieve excellent reproduction of three-dimensional sound?elds using such methods;however,it has been demonstrated that this requires precise measurement of each listener’s individual HRTF’s[20].This seemingly funda-mental requirement that derives from inherent physiological and cognitive characteristics of the human ear–brain inter-face has rendered such systems impractical for widespread use.Current research in this area is focused on achiev-ing good localization performance while using synthetic (nonindividualized)HRTF’s derived through averaging or modeling or based on the HRTF’s of subjects that have been determined to be“good localizers”[21],[22].In his review of the challenges in3-D audio implementations, Begault[23]points out that there are currently three major barriers to successful implementation of such systems:1) psychoacoustic errors such as front–back reversals typical in headphone-based systems,2)large amounts of data required to represent measured HRTF’s accurately,and3) frequency-and phase-response errors that arise from mis-matches between nonindividualized and measured HRTF’s. It should be noted that front–back reversals can be reduced if the listener is allowed to move his head and that the lack of externalization experienced with headphone listening can be alleviated with appropriate use of reverberation.
A fourth challenge arises from technological limitations of current computing systems.One capability that we envision for immersive audio systems is the simulation of room acoustics and listener characteristics for interactive, virtual-,and augmented-reality applications.In addition to the computational requirements for photorealistic rendering of visual images,the synthesis of such acoustical environ-ments requires computation of the binaural room response and subsequent convolution with the HRTF’s of the listener in real time as the listener moves around the room.Typical impulse response duration is3s,which,when sampled at48kHz,requires a processor capable of operating at
944PROCEEDINGS OF THE IEEE,VOL.86,NO.5,MAY1998
more than13G?ops/channel[24].This problem can be circumvented using special-purpose hardware or hybrid block fast Fourier transform/direct convolution methods [25].2The main goal is to reduce the number of operations for such computations,thus making them suitable for real-time interactive applications.
B.Spatial Audio Rendering
A critical issue in the implementation of immersive audio is the reproduction of3-D sound?elds that pre-serve the desired spatial location,frequency response,and dynamic range.There are two general methods for3-D audio rendering that can be categorized as“head related”based on headphone reproduction and“nonhead related”based on loudspeaker reproduction[19].A hybrid category, called transaural stereo,also exists that allows loudspeaker rendering of head-related signals.It should be noted that there are other methods for three-dimensional sound-?eld capture and synthesis such as ambisonics[26]and wave-?eld synthesis[27],[28],but these will not be examined in this paper.
Nonheadroom-related methods typically use multiple loudspeakers to reproduce multiple matrixed or discrete channels.Such systems can convey precisely localized sound images that are primarily con?ned to the horizontal plane and diffuse(ambient)sound to the sides of and behind the listener.In addition to the left and right loudspeakers,they make use of a center loudspeaker that helps create a solidly anchored center-stage sound image,as well as two loudspeakers for the ambient surround sound ?eld.The most prevalent systems currently available to consumers are based on the formats developed by Dolby for?lm sound,including Pro Logic(four-channel matrixed encoded on two channels)and Dolby Digital(5.1-channel discrete based on the AC-3compression scheme).1Other 5.1-channel schemes include DTS Digital Surround3and MPEG-2.Multichannel systems were designed primarily for authentic reproduction of sound associated with movies but have recently started to be used for music recordings and games on CD-ROM.The5.1-channel Dolby Digital system was adopted in the U.S.standard of the upcoming advanced(high-de?nition)television system[29].The design requirements for such loudspeaker-based systems include uniform audience coverage,accurate localization relative to visual images on the screen,diffuse rendering of ambient sounds,and capability for reproduction of the wide (up to105dB)dynamic range present in?lm soundtracks. Head-related binaural recording,or dummy-head stereophony,methods attempt to accurately reproduce at each eardrum of the listener the sound pressure generated by a set of sources and their interactions with the acoustic environment[30].Such recordings can be made with specially designed probe microphones that are inserted in the listener’s ear canal or by using a dummy-head microphone system that is based on average human 2See Lake DSP at https://www.doczj.com/doc/6918137144.html,.
3See https://www.doczj.com/doc/6918137144.html,/.characteristics.Sound recorded using binaural methods is then reproduced through headphones that deliver the desired sound to each ear.It was concluded from early experiments that in order to achieve the desired degree of realism using binaural methods,the required frequency-response accuracy of the transfer function
was
,,
and
speaker
speaker
(speaker)the sound pressure delivered by each loudspeaker.To accurately reproduce the desired binaural signal at each ear,the input
signals
Fig.5.Transfer functions associated with a loudspeaker sound-rendering system.To deliver the correct binaural sound, it is necessary to pre?lter the signal to the loudspeakers so that the
cross-talk terms H LR and H RL are cancelled during reproduction.
ear speaker
ear speaker
ear ear
ear ear
Fig.6.The time-domain response of a loudspeaker system includes the direct sound as well as the sound due to multiple re?ections from the local acoustical environment.Psychoacoustic evidence indicates that in order for these re?ections not to be perceived,their spectrum level should be 15dB below the level of the direct
sound.
Fig.7.Frequency-response problems that arise in the low frequencies due to standing-wave buildup in small rooms and in higher frequencies due to interactions with elements in the local acoustical environment (e.g.,CRT screen,table top,untreated walls).
midbass frequencies.Furthermore,even if it were possible to place the loudspeakers far away from large re?ecting surfaces,this would only solve the problem for middle and high frequencies.Low-frequency room modes do not depend on surfaces in the local acoustical environment but rather on the physical size of the room.These modes produce standing waves that give rise to large variations in frequency response (Fig.7).Such amplitude and phase distortions can completely destroy carefully designed 3-D audio reproduction that relies on the transaural techniques described above.
B.Design Requirements
To circumvent these limitations,a set of solutions has been developed for single-listener desktop reproduction that delivers sound quality equivalent to a calibrated dubbing stage [43].These solutions include direct-path dominant design and correct low-frequency response.
Based on our current understanding of psychoacoustic principles,it is possible to combine such cues to place the listener in a direct sound ?eld that is dominant over the re?ected and reverberant sound.The design considerations
KYRIAKAKIS:LIMITATIONS OF IMMERSIVE AUDIO SYSTEMS
947
Fig.8.A properly designed direct-path dominant system that compensates for frequency anomalies
produces a much?atter frequency response.Frequencies below100Hz are reproduced with a
separate subwoofer(response not shown)that is placed at a known distance from the listener to
alleviate anomalies from standing waves.
for this direct-path dominant design include compensation
of the physical(re?ection and diffraction)effects of the
video/computer monitor that extends the loudspeaker baf?e
as well as the large re?ecting surface on which the computer
keyboard typically rests.The distortions that arise from am-
plitude and phase anomalies are eliminated,and this results
in a listening experience that is dramatically different than
what is achievable through traditional near-?eld monitoring
methods(Fig.8).
Standing waves associated with the acoustics of small
rooms give rise to fundamental limitations in the quality
of reproduced sound,particularly in the uniformity of low-
frequency response.Variations in this frequency regime can
be as large as
s apart),thereby giving rise to acoustic cross
talk[Fig.1(b)].These time differences,combined with
re?ection and diffraction effects caused by the head,lead to
frequency-response anomalies that are perceived as a lack
of clarity[48].
This problem can be solved by adding a cross-talk
cancellation?lter(as described above in the description
of transaural methods)to the signal of each loudspeaker.
While this solution may be satisfactory for the static case,
as soon as the listener moves even slightly,the conditions
948PROCEEDINGS OF THE IEEE,VOL.86,NO.5,MAY1998
Fig.9.Desktop sound system with vision-based head tracking.In this early prototype,the time difference of arrival at the two ears is adjusted in real time as the listener moves in the plane parallel to the loudspeakers.Current research is focused on tracking,pose estimation (for head rotations),and pinna shape recognition for real-time cross-talk cancellation and individualized HRTF synthesis.
for cancellation are no longer met,and the phantom im-age moves toward the closest loudspeaker because of the precedence effect.In order,therefore,to achieve the highest possible quality of sound for a nonstationary listener and preserve the spatial information in the original material,it is necessary to know the precise location of the listener relative to the loudspeakers [47],[49],[50].In the section below,we describe an experimental system that incorpo-rates a novel listener-tracking method in order to overcome the dif?culties associated with two-ear listening as well as the technological limitations imposed by loudspeaker-based desktop audio systems.
VI.F UTURE R ESEARCH D IRECTIONS
A.Vision-Based Methods for Listener Tracking
Computer vision has historically been considered problematic,particularly for tasks that require object recognition.Up to now,the complexity of vision-based approaches has prevented their being incorporated into desktop-based integrated media systems.Recently,however,von der Malsburg’s Laboratory of Computational and Biological Vision at the University of Southern California (USC)has developed a vision architecture that is capable of recognizing the identity,spatial position (pose),facial expression,gesture identi?cation,and movement of a human subject in real time .
This highly versatile architecture integrates a broad va-riety of visual cues in order to identify the location of a person’s head within the image.Object recognition is achieved through pattern-based analysis that identi?es con-vex regions with skin color that are usually associated with the human face and through a stereo algorithm that determines the disparities among pixels that have been moving [51].This pattern-recognition approach is based on the elastic graph matching method that places graph nodes at appropriate ?ducial points of the pattern [52].A set of features is extracted at each graph node corresponding to the amplitudes of complex Gabor wavelets.The key advantage of this method is that a new pattern (face or ear)can be recognized on the basis of a small number of example images (10–100).For audio applications,in which
the system must remember the last position of a listener that may have stopped moving,a hysteresis mechanism is used to estimate the current position and velocity of the head with a linear predictive ?lter.
While there are several alternative methods for track-ing humans (e.g.,magnetic,ultrasound,infrared,laser),they typically are based on tethered operations or require arti?cial ?ducial markings (e.g.,colored dots,earrings)to be worn by the user.Furthermore,these methods do not offer any additional functionality to match what can be achieved with vision-based methods (e.g.,face and expression recognition,ear classi?cation).
B.Desktop Audio System with Head Tracking
A novel multichannel desktop audio system that meets all the design requirements and acoustical considerations described above has been developed by Holman of TMH Corporation 4in collaboration with the Immersive Audio Laboratory at USC’s Integrated Media Systems Center (IMSC).5This system uses two loudspeakers that are po-sitioned on the sides of a video monitor at a distance of 45cm from each other and 50cm from the listener’s ears (Fig.9).The seating position height is adjusted so that the listener’s ears are at the tweeter level of the loudspeakers (117cm from the ?oor),thus eliminating any colorations in the sound due to off-axis lobing.We have also incorporated the vision-based tracking algorithm described above using a standard video camera connected to an SGI Indy workstation.This tracking system provides us with the coordinates of the center of the listener’s head relative to the loudspeakers and is currently capable of operating at 10frames/s with a 3%accuracy.
In this single-camera system,it is possible to track listener movement that is con?ned in a plane parallel to loudspeakers and at a ?xed distance from them.When the listener is located at the exact center position (the sweet spot),sound from each loudspeaker arrives at the corresponding ear at the exact same time (i.e.,with zero ipsilateral time delay).At any other position of the listener
4See https://www.doczj.com/doc/6918137144.html,.5See
https://www.doczj.com/doc/6918137144.html,.
KYRIAKAKIS:LIMITATIONS OF IMMERSIVE AUDIO SYSTEMS
949
in this plane,there is a relative time difference of arrival between the sound signals from each loudspeaker.To maintain proper stereophonic perspective,the ipsilateral time delay must be adjusted as the listener moves relative to the loudspeakers.The head coordinates provided from the tracking algorithm are used to determine the necessary time-delay adjustment.This information is processed by a 32-b DSP processor board(ADSP-2106x SHARC)resident in a Pentium-II PC.In this early version of our system,the DSP board is used to delay the sound from the loudspeaker that is closest to the listener so that sound arrives with the same time difference as if the listener were positioned in the exact center between the loudspeakers.In other words, we have demonstrated stereophonic reproduction with an adaptively optimized sweet spot.
We are currently in the process of identifying the bottle-necks of both the tracking and the audio signal-processing algorithms and integrating both into a single,PC-based platform for real-time operation.Furthermore,we are ex-panding the capability of the current single-camera system to include a second camera in a stereoscopic con?guration that will provide distance(depth)information.
C.Pinna Classi?cation for Enhanced Sound Localization Immersive audio systems based on averaged HRTF’s suffer from serious drawbacks.To map the entire three-dimensional auditory space requires a large number of tedious and time-consuming measurements,which is very dif?cult to do with human subjects.A further,and perhaps insurmountable,complication arises from the fact that this process must be repeated for every listener in order to produce accurate https://www.doczj.com/doc/6918137144.html,st,discrete point measurements represent a quantization of3-D space that is inherently continuous,thus requiring sophisticated interpolation al-gorithms that are computationally intensive and can give rise to errors[53].Several methods have been proposed to overcome such limitations.Functional HRTF represen-tations that make use of models to represent HRTF’s have been proposed[54]–[56];however,most are not suitable for real-time applications because they require signi?cant computational resources.
There is signi?cant evidence to suggest that the identi-?cation and incorporation of pinna physical characteristics may be a key factor limiting the development of seamless immersive audio systems.The human pinna is a rather complicated structure that for many years was considered to be a degenerate remain from past evolutionary forms.It was assumed to be a sound-collecting horn whose purpose was to direct sound into the ear canal.If this were true,then its physical dimensions would limit its role to a collector of high frequencies(short wavelengths).Experimental results, however,have shown that the pinna is a much more sophisticated instrument[17],[54],[57],[58].The pinna folds act as miniature re?ectors that create small time delays,which in turn give rise to comb-?ltering effects in the frequency domain[59].Also,the pinna is asymmetric relative to the opening of the ear canal.These ridges are arranged in such a way as to optimally translate a change in angle of the incident sound into a change in the pattern of re?ections.It has been demonstrated[57]that the human ear–brain interface can detect delay differences as short as
7 in azimuth(directly behind the listener),the pinna also acts as a low-pass?lter,thus providing additional localization cues.
To understand the fundamental limitations imposed by such pinna transformations,the IMSC Immersive Audio Laboratory,in collaboration with the USC Laboratory for Computational Vision,is developing a novel method for classi?cation and cross comparison of pinna characteristics. We are currently in the process of implementing a data base of pinna images and associated measured directional characteristics(HRTF’s).A picture of the pinna from every new listener allows us to select the HRTF from this data base that corresponds to the ear whose pinna shape is closest to the new ear.The algorithm that will be used for this identi?cation is a modi?ed version of the face-recognition algorithm described above.Initial results have shown successful matching of ears from unknown listeners to those already in our data base,including two arti?cial ears from the KEMAR dummy-head system.A planned extension of this matching method will select characteristics from several stored ears that best match the corresponding characteristics of the new pinna.An appropriate set of weighting factors will then be determined to form a synthetic HRTF that closely resembles that of the new listener.It is important to note that this method offers signi?cant advantages over previous model-based attempts because it can be performed very fast and with minimum computational overhead.
A CKNOWLEDGMENT
The author wishes to thank Prof.T.Holman of the USC IMSC and TMH Corporation for his continued guidance and support,as well as Prof.C.von der Malsburg and Dr.H. Neven from the USC Laboratory for Computational Vision for the development and integration of the vision-based head-tracking and pinna classi?cation algorithms.
R EFERENCES
[1]Y.Ando,Concert Hall Acoustics.Berlin,Germany:Springer-
Verlag,1985.
[2]L.Beranek,Concert and Opera Halls:How They Sound.
Woodbury,NY:Acoustical Society of America,1996.
[3]S.Komiyama,“Subjective evaluation of angular displacement
between picture and sound directions for HDTV sound sys-tems,”J.Audio Eng.Soc.,vol.37,pp.210–214,1989.
[4]M.R.Schroeder and B.S.Atal,“Computer simulation of sound
transmission in rooms,”in IEEE Int.Conv.Rec.,1963,vol.7.
[5]M.Kleiner, B.-I.Dalenback,and P.Svensson,“Auraliza-
tion—An overview,”J.Audio Eng.Soc.,vol.41,pp.861–945, 1993.
[6] A. D.Blumlein,“Improvements in and relating to sound-
transmission,sound-recording and sound-reproducing systems,”
U.K.Patent394325,1931.
[7]H.Fletcher,“Auditory perspective—Basic requirements,”Elect.
Eng.,vol.53,pp.9–11,1934.
[8]W.B.Snow,“Basic principles of stereophonic sound,”SMPTE
J.,vol.61,pp.567–589,1953.
[9]J.C.Steinberg and W.B.Snow,“Physical factors,”Bell Syst.
Tech.J.,vol.13,pp.245–258,1934.
950PROCEEDINGS OF THE IEEE,VOL.86,NO.5,MAY1998
[10]P.Scheiber,“Quadrasonic sound system,”U.S.Patent
3632886,1973.
[11],“Multidirectional sound system,”U.S.Patent3746792,
1973.
[12] D.H.Cooper and T.Shiga,“Discrete-matrix multichannel
stereo,”J.Audio Eng.Soc.,vol.20,pp.346–360,1972. [13]G.Theile and G.Plenge,“Localization of lateral phantom
sources,”J.Audio Eng.Soc.,vol.25,pp.196–199,1977.
[14]T.Holman,“Channel crossing,”Studio Sound,pp.40–42,1996.
[15]S.P.Thompson,“On the function of the two ears in the
perception of space,”Philos.Mag.,vol.13,pp.406–416,1882.
[16]J.W.Strutt and L.Rayleigh,“On the perception of sound
direction,”Philos.Mag.,vol.13,pp.214–232,1907.
[17]J.Blauert,Spatial Hearing:The Psychophysics of Human Sound
Localization,revised ed.Cambridge,MA:MIT Press,1997.
[18]K.Genuit,“Ein modell zur beschreibung von
au?enohr′u bertragungseigenschaften,”Ph.D.dissertation, RWTH Aachen,Germany,1984.
[19]H.W.Gierlich,“The application of binaural technology,”Appl.
Acoust.,vol.36,pp.219–243,1992.
[20] F.L.Wightman and D.J.Kistler,“Headphone simulation of
free-?eld listening:Psychophysical validation,”J.Acoust.Soc.
Amer.,vol.85,pp.868–878,1989.
[21] E.M.Wenzel,M.Arruda,and D.J.Kistler,“Localization using
nonindividualized head-related transfer functions,”J.Acoust.
Soc.Amer.,vol.94,pp.111–123,1993.
[22] D.R.Begault and E.M.Wenzel,“Headphone localization of
speech,”Human Factors,vol.35,pp.361–376,1993. [23],“Challenges to the successful implementation of3-D
sound,”J.Audio Eng.Soc.,vol.39,pp.864–870,1991. [24]H.Lehnert and J.Blauert,“Principles of binaural room simu-
lation,”Appl.Acoust.,vol.36,pp.259–291,1992.
[25]W.G.Gardner,“Ef?cient convolution without input–output
delay,”J.Audio Eng.Soc.,vol.43,pp.127–136,1995. [26]M.A.Gerzon,“Ambisonics in multichannel broadcasting and
video,”J.Audio Eng.Soc.,vol.33,pp.859–871,1985. [27] A.J.Berkhout,D.de Vries,and P.V ogel,“Acoustics control
by wave?eld synthesis,”J.Acoust.Soc.Amer.,vol.93,p.2764, 1993.
[28]M.M.Boone,E.N.G.Verheijen,and P.F.von Tol,“Spatial
sound-?eld reproduction by wave?eld synthesis,”J.Audio Eng.
Soc.,vol.43,p.1003,1995.
[29] D.J.Meares,“Multichannel sound systems for HDTV,”Appl.
Acoust.,vol.36,pp.245–257,1992.
[30]H.Moller,“Fundamentals of binaural technology,”Appl.
Acoust.,vol.36,pp.171–218,1992.
[31]J.Blauert and https://www.doczj.com/doc/6918137144.html,ws,“Group delay distortions in elec-
troacoustical systems,”J.Acoust.Soc.Amer.,vol.63,pp.
1478–1483,1978.
[32]H.-J.Platte,https://www.doczj.com/doc/6918137144.html,ws,and H.v.H¨o vel,“Apparatus for the exact
reproduction of ear input signals(in German),”Fortschritte der Akustik,vol.DAGA’75,pp.361–364,1975.
[33] F.L.Wightman and D.J.Kistler,“Headphone simulation of
free-?eld listening:stimulus synthesis,”J.Acoust.Soc.Amer., vol.85,pp.858–867,1989.
[34]H.Moller,C.B.Jensen,and D.Hammershoi,“Design criteria
for headphones,”J.Audio Eng.Soc.,vol.43,pp.218–232,1995.
[35] B.B.Bauer,“Stereophonic earphones and binaural loudspeak-
ers,”J.Audio Eng.Soc.,vol.9,pp.148–151,1961.
[36]P.Damaske and V.Mellert,“A procedure for generating
directionally accurate sound images in the upper half-space using two loudspeakers,”Acustica,vol.22,pp.154–162,1969.
[37]P.Damaske,“Head related two channel stereophony with
loudspeaker reproduction,”J.Acoust.Soc.Amer.,vol.50,pp.
1109–1115,1971.
[38] D.H.Cooper and J.L.Bauck,“Prospects for transaural
recording,”J.Audio Eng.Soc.,vol.37,pp.3–19,1989. [39]J.Bauck and D.H.Cooper,“Generalized transaural stereo and
applications,”J.Audio Eng.Soc.,vol.44,pp.683–705,1996.
[40]S.Mehrgard and V.Mellert,“Transformation characteristics of
the external human ear,”J.Acoust.Soc.Amer.,vol.51,pp.
1567–1576,1977.
[41] F.E.Toole,“Loudspeaker measurements and their relation-
ship to listener preferences,”J.Audio Eng.Soc.,vol.34,pp.
227–235,1986.[42]S.Bech,“Perception of timbre of reproduced sound in small
rooms:In?uence of room and loudspeaker position,”J.Audio Eng.Soc.,vol.42,pp.999–1007,1994.
[43]T.Holman,“Monitoring sound in the one-person environment,”
SMPTE J.,vol.106,pp.673–680,1997.
[44]R.Walker,“Room modes and low-frequency responses in small
enclosures,”Audio Eng.Soc.,preprint no.4194,1994. [45]T.Holman,“Report on mixing studios sound quality,”J.Jpn.
Audio Soc.,1994.
[46] F.E.Toole,“Subjective measurements of loudspeaker sound
quality and listener performance,”J.Audio Eng.Soc.,vol.33, pp.2–32,1985.
[47] C.Kyriakakis and T.Holman,“High quality audio for the desk-
top,”J.Visual Commun.Image Representation,to be published.
[48]T.Holman,“New factors in sound for cinema and television,”
J.Audio Eng.Soc.,vol.39,pp.529–539,1991.
[49]T.Holman and C.Kyriakakis,“Acoustics and psychoacoustics
of desktop sound systems,”in Proc.Ann.Meeting Acoustical Society of America,San Diego,CA,1997,p.3092.
[50]W.G.Gardner,“Head-tracked3-D audio using loudspeakers,”
in Proc.WASP AA,New Paltz,NY,1997.
[51]O.Groetenherdt,“Video-based detection of heads using motion
and stereo vision(in German),”in Institute for Neuroinformat-ics.Bochum,Germany:Univ.of Bochum,1997.
[52]L.Wiskott,J.M.Fellous,N.Krueger,and C.von der Malsburg,
“Face recognition by elastic bunch graph matching,”Institute for Neuroinformatics,Tech.Rep.no.8,1996.
[53] F.L.Wightman,D.J.Kistler,and M.Arruda,“Perceptual
consequences of engineering compromises in synthesis of vir-tual auditory objects,”J.Acoust.Soc.Amer.,vol.101,pp.
1050–1063,1992.
[54] D.W.Batteau,“The role of the pinna in human localization,”
Proc.Royal Soc.London,vol.B168,pp.158–180,1967. [55] A.J.Watkins,“The monaural perception of azimuth:A synthe-
sis approach,”in Localization of Sound:Theory and Applica-tions,R.W.Gatehouse,Ed.Groton,CT:Amphora,1979. [56]J.Chen,B.D.Van Veen,and K.E.Hecox,“A spatial feature
extraction and regularization model for the head-related transfer function,”J.Acoust.Soc.Amer.,vol.97,pp.439–452,1995.
[57]J.Hebrank and D.Wright,“Spectral cues used in the local-
ization of sound sources in the median plane,”J.Acoust.Soc.
Amer.,vol.56,pp.1829–1834,1974.
[58]S.S.Stevens and E.B.Newman,“The localization of actual
sources of sound,”Amer.J.Physiology,vol.48,pp.297–306, 1936.
[59] C.A.P.Rodgers,“Pinna transformations and sound reproduc-
tion,”J.Audio Eng.Soc.,vol.29,pp.226–234,
1981.
Chris Kyriakakis(Member,IEEE)was born
in Thessaloniki,Greece,in1963.He received
the B.S.degree in electrical engineering from
the California Institute of Technology,Pasadena,
in1985and the M.S.and Ph.D.degrees in
electrical engineering from the University of
Southern California(USC),Los Angeles,in
1987and1993,respectively.
He currently is an Assistant Professor in the
Electrical Engineering Department,USC,and
an Investigator in the Integrated Media Systems Center(IMSC),a National Science Foundation Engineering Research Center at USC.He is the head of the IMSC Immersive Audio Laboratory and is currently involved in the development of algorithms,systems,and architectures for immersive audio.These efforts include audio signal pro-cessing for accurate spatial rendering of sound at the desktop environment as well as experimental investigation of human listening characteristics.He is involved in several collaborative efforts within the IMSC,including the incorporation of a vision-based system to track a listener’s head for precise, spatially accurate rendering of sound in a three-dimensional immersive audio environment as well as implementation of microphone arrays for robust speaker identi?cation and tracking.
KYRIAKAKIS:LIMITATIONS OF IMMERSIVE AUDIO SYSTEMS951
[摘要] 计算机网络安全建设是涉及我国经济发展、社会发展和国家安全的重大问题。本文结合网络安全建设的全面信息，在对网络系统详细的需求分析基础上，依照计算机网络安全设计目标和计算机网络安全系统的总体规划，设计了一个完整的、立体的、多层次的网络安全防御体系。 [关键词] 网络安全方案设计实现 一、计算机网络安全方案设计与实现概述 影响网络安全的因素很多，保护网络安全的技术、手段也很多。一般来说，保护网络安全的主要技术有防火墙技术、入侵检测技术、安全评估技术、防病毒技术、加密技术、身份认证技术，等等。为了保护网络系统的安全，必须结合网络的具体需求，将多种安全措施进行整合，建立一个完整的、立体的、多层次的网络安全防御体系，这样一个全面的网络安全解决方案，可以防止安全风险的各个方面的问题。 二、计算机网络安全方案设计并实现 1.桌面安全系统 用户的重要信息都是以文件的形式存储在磁盘上，使用户可以方便地存取、修改、分发。这样可以提高办公的效率，但同时也造成用户的信息易受到攻击，造成泄密。特别是对于移动办公的情况更是如此。因此，需要对移动用户的文件及文件夹进行本地安全管理，防止文件泄密等安全隐患。 本设计方案采用清华紫光公司出品的紫光S锁产品，“紫光S锁”是清华紫光“桌面计算机信息安全保护系统”的商品名称。紫光S锁的内部集成了包括中央处理器（CPU）、加密运算协处理器（CAU）、只读存储器（ROM），随机存储器（RAM）、电可擦除可编程只读存储器（E2PROM）等，以及固化在ROM内部的芯片操作系统COS（Chip Operating Sys tem）、硬件ID号、各种密钥和加密算法等。紫光S锁采用了通过中国人民银行认证的Sm artCOS，其安全模块可防止非法数据的侵入和数据的篡改，防止非法软件对S锁进行操作。 2.病毒防护系统 基于单位目前网络的现状，在网络中添加一台服务器，用于安装IMSS。
#pragma code_seg 格式如： #pragma code_seg( [ [ { push | pop}, ] [ identifier, ] ] [ "segment-name" [, "segment-class" ] ) 该指令用来指定函数在.obj文件中存放的节,观察OBJ文件可以使用VC自带的dumpbin命令行程序,函数在.obj文件中默认的存放节为.text节，如果code_seg 没有带参数的话,则函数存放在.text节中。 push (可选参数)将一个记录放到内部编译器的堆栈中,可选参数可以为一个标识符或者节名 pop(可选参数)将一个记录从堆栈顶端弹出,该记录可以为一个标识符或者节名identifier(可选参数)当使用push指令时,为压入堆栈的记录指派的一个标识符,当该标识符被删除的时候和其相关的堆栈中的记录将被弹出堆栈 "segment-name" (可选参数)表示函数存放的节名 例如: //默认情况下,函数被存放在.text节中 void func1() {// stored in .text } //将函数存放在.my_data1节中 #pragma code_seg(".my_data1") void func2() {// stored in my_data1 } //r1为标识符,将函数放入.my_data2节中 #pragma code_seg(push, r1, ".my_data2") void func3() {// stored in my_data2 } int main() { } 例如 #pragma code_seg(“PAGE”) 作用是将此部分代码放入分页内存中运行。 #pragma code_seg() 将代码段设置为默认的代码段 #pragma code_seg("INIT") 加载到INIT内存区域中，成功加载后，可以退出内存
校园网络与信息安全管理办法 桑梓镇辛撞中心小学 2018.9
校园网络与信息安全管理办法 学校校园网是为教学及学校管理而建立的计算机信息网络，目的在于利用先进实用的计算机技术和网络通信技术，实现校园内计算机互联、资源共享，并为师生提供丰富的网上资源。为了保护校园网络系统的安全、促进学校计算机网络的应用和发展，保证校园网络的正常运行和网络用户的使用权益，更好的为教育教学服务，特制定如下管理办法。 第一章总则 1、本管理制度所称的校园网络系统，是指由校园网络设备、配套的网络线缆设施、网络服务器、工作站、学校办公及教师教学用计算机等所构成的，为校园网络应用而服务的硬件、软件的集成系统。 2、校园网络的安全管理，应当保障计算机网络设备和配套设施的安全，保障信息的安全和运行环境的安全，保障网络系统的正常运行，保障信息系统的安全运行。 3、按照“合法合规，遵从标准”的原则开展网络与信息安全管理工作，网络及信息安全管理领导小组负责相应的网络安全和信息安全工作，定期对相应的网络用户进行有关信息安全和网络安全教育并对上网信息进行审查和监控。 4、所有上网用户必须遵守国家有关法律、法规，严格执行安全保密制度，并对所提供的信息负责。任何单位和个人不得利用联网计算机从事危害校园网及本地局域网服务器、工作站的活动。 5、进入校园网的全体学生、教职员工必须接受并配合国家有关部门及学校依法进行的监督检查，必须接受学校校园网络及信息安全管理领导小组进行的网络系统及信息系统的安全检查。
6、使用校园网的全体师生有义务向校园网络及信息安全管理领导小组和有关部门报告违法行为和有害信息。 第二章网络安全管理细则 1、网络管理中心机房及计算机网络教室要装置调温、调湿、稳压、接地、防雷、防火、防盗等设备，管理人员应每天检查上述设备是否正常，保证网络设备的安全运行，要建立完整、规范的校园网设备运行情况档案及网络设备账目，认真做好各项资料(软件)的记录、分类和妥善保存工作。 2、校园网由学校信息中心统一管理及维护。连入校园网的各部门、处室、教室和个人使用者必须严格使用由信息中心分配的IP地址，网络管理员对入网计算机和使用者进行及时、准确登记备案，由信息中心负责对其进行监督和检查。任何人不得更改IP及网络设置，不得盗用IP地址及用户帐号。 3、与校园网相连的计算机用户建设应当符合国家的有关标准和规定，校园内从事施工、建设，不得危害计算机网络系统的安全。 4、网络管理员负责全校网络及信息的安全工作，建立网络事故报告并定期汇报，及时解决突发事件和问题。校园网各服务器发生案件、以及遭到黑客攻击后，信息中心必须及时备案并向公安机关报告。 5、网络教室及相关设施未经校领导批准不准对社会开放。 6、按照信息安全等级保护工作规定，完成定级、备案等工作，留存安全审核日志。校园网中对外发布信息的Web服务器中的内容必须经领导审核，由负责人签署意见后再由专人（信息员）发布。新闻公布、公文发布权限要经过校领导的批准。门户网站不得链接企业网站，不得发布商业广告，不得在网页中设置或植入商品、商业服务的二维码。
校园网络安全设计方案 10网工2班组员：张婵、张茜、张越、张喻博、赵子龙、祝美意、杨越峦、张力 随着因特网的迅速发展，校园网的建设日益普遍。而在高校中，如何能够保证校园网络的安全运行，同时又能提供丰富的网络资源，达到办公、教学及学生上网的多种需求已成为了一个难题。校园网络的安全不仅有来自外部的攻击，还有内部的攻击。所以，在校园网建设中使用安全技术是刻不容缓的。现从防火墙、VPN、防病毒、入侵检测和防御系统、上网行为管理和用户审计系统、数据备份系统、主页防篡改、网络安全管理制度几个方面，设计我校的网络安全方案。 防火墙:防火墙是一种将内部网和公众网分开的方法。它能限制被保护的网络与与其他网络之间进行的信息存取、传递操作。 防火墙的概念：通常防火墙是指部署在不同网络或网络安全域之间的一系列部件组合，是一种有效的网络安全策略。防火墙提供信息安全服务，设置在被保护内部网络的安全与不安全的外部网络之间，其作用是阻断来自外部的、针对内部网络的入侵和威胁，保护内部网络的安全。它是不同网络或网络安全域之间信息的唯一出入口，根据安全策略控制出入网络的信息流，并且本身具有较强的抗攻击能力。 防火墙的分类：按软件与硬件的形式，防火墙分为软件防火墙、硬件防火墙和芯片防火墙；按防火墙的技术，总体分为包过滤型和应用代理型两大类；按防火墙的结构分为单一主机防火墙、路由器集成式防火墙、分布式防火墙；按防火墙的部署位置分为边界防火墙、个人防火墙、混合防火墙。 防火墙的安全策略：（1）所有从内到外和从外到内的数据包都必须经过防火墙（2）只有被安全策略允许的数据包才能通过防火墙（3）防火墙本身要有预防入侵的功能（4）默认禁止所有服务，除非是必须的服务才被允许 防火墙的设计：（1）保障校园内部网主机的安全，屏蔽内部网络，禁止外部网用户连接到内部网（2）只向外部用户提供HTTP、SMTP和POP等有限的服务（3）向内部记账用户提供所有Internet服务，但一律通过代理服务器(4)禁止访问黄色、反动网站(5)要求具备防IP 地址欺骗和IP地址盗用功能(6)要求具备记账和审计功能，能有效记录校园网的一切活动。 校园网络在设置时应从下面几个方面入手：（1）入侵检测：具有黑客普通攻击的实时检测技术。实时防护来自IP Source Routing、IP Spoofing、SYN flood、IC-MP flood、UDP flood、Ping ofDeath、拒绝服务和许多其它攻击。并且在检测到有攻击行为时能通过电子邮件或其它方式通知系统管理员。（2）工作模式选择：目前市面上的防火墙都会具备三种不同的工作模式，路由模式、NAT模式和透明模式。我们选择的是透明模式，防火墙过滤通过防火墙的封包，而不会修改数据包包头中的任何源或目的地的信息。所有接口运行起来都像是同一网络中的一部分。此时防火墙的作用更像是Layer2（第二层）交换机或桥接器。在透明模式下，接口的IP地址被设置为0.0.0.0, 防火墙对于用户来说是可视或透明的。（3）策略设置：防火墙可以提供具有单个进入和退出点的网络边界。由于所有信息流都必须通过此点，因此可以筛选并引导所有通过执行策略组列表产生的信息流。策略能允许、拒绝、加密、认证、排定优先次序、调度以及监控尝试从一个安全段流到另一个安全段的信息流。可以决定哪些用户和信息能进入和离开以及它们进入和离开的时间和地点。（4）管理界面：管理一个防火墙的方法一般来说有两种：图形化界面和命令行界面，我们选择为通过web方式和java等程序编写的图形化界面进行远程管理。（5）内容过滤：面对当前互联网上的各种有害信息，我们的防火墙还增加了URL阻断、关键词检查、Java Ap-ple、ActiveX和恶意脚本过滤等。（6）防火墙的性能考虑：防火墙的性能对于一个防火墙来说是至关重要的，它决定了每秒钟可能通过防火墙的最大数据流量，以bps为单位，从几十兆到几百兆不等。千兆防火墙还
#pragma预处理命令 #pragma可以说是C++中最复杂的预处理指令了，下面是最常用的几个#pragma 指令： #pragma comment(lib,"XXX.lib") 表示链接XXX.lib这个库，和在工程设置里写上XXX.lib的效果一样。 #pragma comment(linker,"/ENTRY:main_function") 表示指定链接器选项/ENTRY:main_function #pragma once 表示这个文件只被包含一次 #pragma warning(disable:4705) 表示屏蔽警告4705 C和C++程序的每次执行都支持其所在的主机或操作系统所具有的一些独特的特点。例如，有些程序需要精确控制数据存放的内存区域或控制某个函数接收的参数。#pragma为编译器提供了一种在不同机器和操作系统上编译以保持C和C++完全兼容的方法。#pragma是由机器和相关的操作系统定义的，通常对每个编译器来说是不同的。 如果编译器遇到不认识的pragma指令，将给出警告信息，然后继续编译。Microsoft C and C++ 的编译器可识别以下指令：alloc_text，auto_inline，bss_seg，check_stack，code_seg，comment，component，conform，const_seg，data_seg，deprecated，fenv_access，float_control，fp_contract，function，hdrstop，include_alias，init_seg，inline_depth，inline_recursion，intrinsic，make_public，managed，message，omp，once，optimize，pack，pointers_to_members，pop_macro，push_macro，region, endregion，runtime_checks，section，setlocale，strict_gs_check，unmanaged，vtordisp，warning。其中conform，init_seg, pointers_to_members，vtordisp仅被C++编译器支持。 以下是常用的pragma指令的详细解释。 1.#pragma once。保证所在文件只会被包含一次，它是基于磁盘文件的，而#ifndef 则是基于宏的。
#pragma data:code 在Keil中为了节省数据存储器的空间，通过“code”关键字来定义一个数组或字符串将被存储在程序存储器中： uchar code buffer[]={0,1,2,3,4,5}; uchar code string[]="Armoric" ; 而这类代码移值到ICCAVR上时是不能编译通过的。我们可以通过"const" 限定词来实现对存储器的分配： #pragma data:code const unsigned char buffer[]={0,1,2,3,4,5}; const unsigned char string[]="Armoric"; #pragma data:data 注意： 《1》使用ICCAVR6.31时，#pragma data :code ;#pragma data:data ; 这些语法时在"data:cod"、"data:data"字符串中间不能加空格，否则编译不能通过。 《2》const 在ICCAVR是一个扩展关键词，它与ANSIC标准有冲突，移值到其它的编译器使用时也需要修改相关的地方。 在ICCAVR中对数组和字符串的五种不同空间分配： const unsigned char buffer[]={0,1,2,3,4,5}; //buffer数组被分配在程序存储区中 const unsigned char string[]="Armoric" ; //stringp字符串被分配在程序存储区中 const unsigned char *pt //指针变量pt被分配在数据存储区中，指向程序存储区中的字符类型数据 unsigned char *const pt //指针变量pt被分配在程序存储区中，指向数据存储区中的字符类型数据 const unsigned char *const pt //指针变量pt被分配在程序存储区，指向程序存储区中的字符类型数据 unsigned char *pt //指针变量pt被分配在数据存储区中，指向数据存储区中的数据 请问#pragma data:code和#pragma data:data是什么意思？ 前者表示：随后的数据将存贮在程序区，即FLASH区，此区只能存贮常量，比如表格之类。
校园网网络安全设计方案 网络技术的高速发展,各种各样的安全问题也相继出现,校园网被“黑”或被病毒破坏的事件屡有发生,造成了极坏的社会影响和巨大的经济损失。维护校园网网络安全需要从网络的搭建及网络 安全设计方面着手。 一、基本网络的搭建。 由于校园网网络特性(数据流量大,稳定性强,经济性和扩充性)和各个部门的要求(制作部门和办公部门间的访问控制),我们采用下列方案: 1. 网络拓扑结构选择:网络采用星型拓扑结构(如图1)。它是目前使用最多,最为普遍的局域网拓扑结构。节点具有高度的独立性,并且适合在中央位置放置网络诊断设备。 2.组网技术选择:目前,常用的主干网的组网技术有快速以太网(100Mbps)、FDDI、千兆以太网(1000Mbps)和ATM(155Mbps/622Mbps)。快速以太网是一种非常成熟的组网技术,它的造价很低,性能价格比很高;FDDI也是一种成熟的组网技术,但技术复杂、造价高,难以升级;ATM技术成熟,是多媒体应用系统的理想网络平台,但它的网络带宽的实际利用率很低;目前千兆以太网已成为一种成熟的组网技术,造价低于ATM网,它的有效带宽比622Mbps的ATM 还高。因此,个人推荐采用千兆以太网为骨干,快速以太网交换到桌面组建计算机播控网络。 二、网络安全设计。 1.物理安全设计为保证校园网信息网络系统的物理安全,除在网络规划和场地、环境等要求之外,还要防止系统信息在空间的扩散。计算机系统通过电磁辐射使信息被截获而失密的案例已经很多,在理论和技术支持下的验证工作也证实这种截取距离在几百甚至可达千米的复原显示技术给计算机系统信息的保密工作带来了极大的危害。为了防止系统中的信息在空间上的扩散,通常是在物理上采取一定的防护措施,来减少或干扰扩散出去的空间信号。正常的防范措施主要在三个方面:对主机房及重要信息存储、收发部门进行屏蔽处理,即建设一个具有高效屏蔽效能的屏蔽室,用它来安装运行主要设备,以防止磁鼓、磁带与高辐射设备等的信号外泄。为提
#pragma的用法 在所有的预处理指令中，#Pragma 指令可能是最复杂的了，它的作用是设定编译器的状态或者是指示编译器完成一些特定的动作。#pragma指令对每个编译器给出了一个方法，在保持与C和C++语言完全兼容的情况下，给出主机或操作系统专有的特征。依据定义， 编译指示是机器或操作系统专有的，且对于每个编译器都是不同的。 其格式一般为: #pragma para。其中para为参数，下面来看一些常用的参数。 1)message 参数 message参数是我最喜欢的一个参数，它能够在编译信息输出窗口中输出相应的信息，这对于源代码信息的控制是非常重要的。其使用方法为： #pragma message("消息文本") 当编译器遇到这条指令时就在编译输出窗口中将消息文本打印出来。 当我们在程序中定义了许多宏来控制源代码版本的时候，我们自己有可能都会忘记有 没有正确的设置这些宏， 此时我们可以用这条指令在编译的时候就进行检查。假设我们希望判断自己有没有在源代码的什么地方定义了_X86这个宏, 可以用下面的方法: #ifdef _X86 #pragma message("_X86 macro activated!") #endif 我们定义了_X86这个宏以后，应用程序在编译时就会在编译输出窗口里显示"_86 macro activated!"。 我们就不会因为不记得自己定义的一些特定的宏而抓耳挠腮了。 (2)另一个使用得比较多的pragma参数是code_seg 格式如： #pragma code_seg( ["section-name" [, "section-class"] ] ) 它能够设置程序中函数代码存放的代码段，当我们开发驱动程序的时候就会使用到 它。 (3)#pragma once (比较常用) 只要在头文件的最开始加入这条指令就能够保证头文件被编译一次，这条指令实际上 在VC6中就已经有了， 但是考虑到兼容性并没有太多的使用它。 (4)#pragma hdrstop 表示预编译头文件到此为止，后面的头文件不进行预编译。BCB可以预编译头文件以 加快链接的速度， 但如果所有头文件都进行预编译又可能占太多磁盘空间，所以使用这个选项排除一些头文
xx校园网网络安全解决方案1 xx校园网网络安全解决方案 校园网网络是一个分层次的拓扑结构，因此网络的安全防护也需采用分层次的拓扑防护措施。即一个完整的校园网网络信息安全解决方案应该覆盖网络的各个层次，并且与安全管理相结合。 一、网络信息安全系统设计原则 1.1满足Internet分级管理需求 1.2需求、风险、代价平衡的原则 1.3综合性、整体性原则 1.4可用性原则 1.5分步实施原则 目前，对于新建网络及已投入运行的网络，必须尽快解决网络的安全保密问题，设计时应遵循如下思想： （1）大幅度地提高系统的安全性和保密性； （2）保持网络原有的性能特点，即对网络的协议和传输具有很好的透明性； （3）易于操作、维护，并便于自动化管理，而不增加或少增加附加操作；
（4）尽量不影响原网络拓扑结构，便于系统及系统功能的扩展； （5）安全保密系统具有较好的性能价格比，一次性投资，可以长期使用； （6）安全与密码产品具有合法性，并便于安全管理单位与密码管理单位的检查与监督。 基于上述思想，网络信息安全系统应遵循如下设计原则： 满足因特网的分级管理需求根据Internet网络规模大、用户众多的特点，对Internet/Intranet信息安全实施分级管理的解决方案，将对它的控制点分为三级实施安全管理。 --第一级：中心级网络，主要实现内外网隔离；内外网用户的访问控制；内部网的监控；内部网传输数据的备份与稽查。 --第二级：部门级，主要实现内部网与外部网用户的访问控制；同级部门间的访问控制；部门网内部的安全审计。 --第三级：终端/个人用户级，实现部门网内部主机的访问控制；数据库及终端信息资源的安全保护。 需求、风险、代价平衡的原则对任一网络，绝对安全难以达到，也不一定是必要的。对一个网络进行实际额研究（包括任务、性能、结构、可靠性、可维护性等），并对网络面临的威胁及可能承担的风险进行定性与定量相结合的分析，然后制定规范和措施，确定本系统的安全策略。
#pragma pack(4) //按4字节对齐，但实际上由于结构体中单个成员的最大占用字节数为2字节，因此实际还是按2字节对齐 typedef struct { char buf[3];//buf[1]按1字节对齐，buf[2]按1字节对齐，由于buf[3]的下一成员word a是按两字节对齐，因此buf[3]按1字节对齐后，后面只需补一空字节 word a; //#pragma pack(4)，取小值为2，按2字节对齐。 }kk; #pragma pack() //取消自定义字节对齐方式 对齐的原则是min(sizeof(word ),4)=2,因此是2字节对齐，而不是我们认为的4字节对齐。 这里有三点很重要: 1.每个成员分别按自己的方式对齐,并能最小化长度 2.复杂类型(如结构)的默认对齐方式是它最长的成员的对齐方式,这样在成员是复杂类型时,可以最小化长度 3.对齐后的结构体整体长度必须是成员中最大的对齐参数的整数倍,这样在处理数组时可以保证每一项都边界对齐 补充一下,对于数组,比如: char a[3];这种,它的对齐方式和分别写3个char是一样的.也就是说它还是按1个字节对齐. 如果写: typedef char Array3[3]; Array3这种类型的对齐方式还是按1个字节对齐,而不是按它的长度. 不论类型是什么,对齐的边界一定是1,2,4,8,16,32,64....中的一个. 声明： 整理自网络达人们的帖子，部分参照MSDN。 作用： 指定结构体、联合以及类成员的packing alignment; 语法： #pragma pack( [show] | [push | pop] [, identifier], n ) 说明： 1，pack提供数据声明级别的控制，对定义不起作用； 2，调用pack时不指定参数，n将被设成默认值； 3，一旦改变数据类型的alignment，直接效果就是占用memory的减少，但是performance会下降； 语法具体分析： 1，show：可选参数；显示当前packing aligment的字节数，以warning message的形式被显示； 2，push：可选参数；将当前指定的packing alignment数值进行压栈操作，这里的栈是the internal compiler stack，同时设置当前的packing alignment为n；如果n没有指定，则将当前的packing alignment数值压栈； 3，pop：可选参数；从internal compiler stack中删除最顶端的record；如果没有指定n，则当前栈顶record即为新的packing alignment 数值；如果指定了n，则n将成为新的packing aligment数值；如果指定了identifier，则internal compiler stack中的record都将被pop 直到identifier被找到，然后pop出identitier，同时设置packing alignment数值为当前栈顶的record；如果指定的identifier并不存在于internal compiler stack，则pop操作被忽略； 4，identifier：可选参数；当同push一起使用时，赋予当前被压入栈中的record一个名称；当同pop一起使用时，从internal compiler stack 中pop出所有的record直到identifier被pop出，如果identifier没有被找到，则忽略pop操作； 5，n：可选参数；指定packing的数值，以字节为单位；缺省数值是8，合法的数值分别是1、2、4、8、16。 重要规则： 1，复杂类型中各个成员按照它们被声明的顺序在内存中顺序存储，第一个成员的地址和整个类型的地址相同； 2，每个成员分别对齐，即每个成员按自己的方式对齐，并最小化长度；规则就是每个成员按其类型的对齐参数（通常是这个类型的大小）和指定对齐参数中较小的一个对齐； 3，结构体、联合体或者类的数据成员，第一个放在偏移为0的地方；以后每个数据成员的对齐，按照#pragma pack指定的数值和这个数据成员自身长度两个中比较小的那个进行；也就是说，当#pragma pack指定的值等于或者超过所有数据成员长度的时候，这个指定值的大小将不产生任何效果； 4，复杂类型（如结构体）整体的对齐是按照结构体中长度最大的数据成员和#pragma pack指定值之间较小的那个值进行；这样当数据成员为复杂类型（如结构体）时，可以最小化长度； 5，复杂类型（如结构体）整体长度的计算必须取所用过的所有对齐参数的整数倍，不够补空字节；也就是取所用过的所有对齐参数中最大的那个值的整数倍，因为对齐参数都是2的n次方；这样在处理数组时可以保证每一项都边界对齐； 对齐的算法：由于各个平台和编译器的不同，现以本人使用的gcc version 3.2.2编译器（32位x86平台）为例子，来讨论编译器对struct 数据结构中的各成员如何进行对齐的。 在相同的对齐方式下，结构体内部数据定义的顺序不同，结构体整体占据内存空间也不同，如下： 设结构体如下定义： struct A { int a; //a的自身对齐值为4，偏移地址为0x00~0x03，a的起始地址0x00满足0x00%4=0;
校园网络安全设计方案 一、安全需求 1.1.1网络现状 随着信息技术的不断发展和网络信息的海量增加，校园网的安全形势日益严峻，目前校园网安全防护体系还存在一些问题，主要体现在：网络的安全防御能力较低，受到病毒、黑客的影响较大，对移动存储介质的上网监测手段不足，缺少综合、高效的网络安全防护和监控手段，削弱了网络应用的可靠性。因此，急需建立一套多层次的安全管理体系，加强校园网的安全防护和监控能力，为校园信息化建设奠定更加良好的网络安全基础。 经调查，现有校园网络拓扑图如下： 1.1.2应用和信息点
1.2.现有安全技术 1．操作系统和应用软件自身的身份认证功能，实现访问限制。 2．定期对重要数据进行备份数据备份。 3．每台校园网电脑安装有防毒杀毒软件。 1.3.安全需求 1．构建涵盖校园网所有入网设备的病毒立体防御体系。 计算机终端防病毒软件能及时有效地发现、抵御病毒的攻击和彻底清除病毒，通过计算机终端防病毒软件实现统一的安装、统一的管理和病毒库的更新。 2. 建立全天候监控的网络信息入侵检测体系 在校园网关键部位安装网络入侵检测系统，实时对网络和信息系统访问的异常行为进行监测和报警。 3. 建立高效可靠的内网安全管理体系 只有解决网络内部的安全问题，才可以排除网络中最大的安全隐患，内网安全管理体系可以从技术层面帮助网管人员处理好繁杂的客户端问题。 4. 建立虚拟专用网(VPN)和专用通道 使用VPN网关设备和相关技术手段，对机密性要求较高的用户建立虚拟专用网。 二．安全设计 1．1设计原则 根据防范安全攻击的安全需求、需要达到的安全目标、对应安全机制所需的安全服务等因素，参照SSE-CMM("系统安全工程能力成熟模型")和ISO17799(信息安全管理标准)等国际标准，综合考虑可实施性、可管理性、可扩展性、综合完备
#pragma指令用法汇总和解析 一. message 参数。 message 它能够在编译信息输出窗 口中输出相应的信息，这对于源代码信息的控制是非常重要的。其使用方法为： #pragma message(“消息文本”) 当编译器遇到这条指令时就在编译输出窗口中将消息文本打印出来。 当我们在程序中定义了许多宏来控制源代码版本的时候，我们自己有可能都会忘记有没有正确的设置这些宏，此时我们可以用这条 指令在编译的时候就进行检查。假设我们希望判断自己有没有在源代码的什么地方定义了_X86这个宏可以用下面的方法 #ifdef _X86 #pragma message(“_X86 macro activated!”) #endif 当我们定义了_X86这个宏以后，应用程序在编译时就会在编译输出窗口里显示“_ X86 macro activated!”。我们就不会因为不记得自己定义的一些特定的宏而抓耳挠腮了 二. 另一个使用得比较多的#pragma参数是code_seg。格式如： #pragma code_seg( [ [ { push | pop}, ] [ identifier, ] ] [ "segment-name" [, "segment-class" ] ) 该指令用来指定函数在.obj文件中存放的节,观察OBJ文件可以使用VC自带的dumpbin命令行程序,函数在.obj文件中默认的存放节 为.text节 如果code_seg没有带参数的话,则函数存放在.text节中 push (可选参数) 将一个记录放到内部编译器的堆栈中,可选参数可以为一个标识符或者节名 pop(可选参数) 将一个记录从堆栈顶端弹出,该记录可以为一个标识符或者节名 identifier (可选参数) 当使用push指令时,为压入堆栈的记录指派的一个标识符,当该标识符被删除的时候和其相关的堆栈中的记录将被弹出堆栈 "segment-name" (可选参数) 表示函数存放的节名 例如: //默认情况下,函数被存放在.text节中 void func1() { // stored in .text } //将函数存放在.my_data1节中 #pragma code_seg(".my_data1") void func2() { // stored in my_data1 } //r1为标识符,将函数放入.my_data2节中 #pragma code_seg(push, r1, ".my_data2") void func3() { // stored in my_data2 } int main() { } 三. #pragma once (比较常用） 这是一个比较常用的指令,只要在头文件的最开始加入这条指令就能够保证头文件被编译一次 四. #pragma hdrstop表示预编译头文件到此为止，后面的头文件不进行预编译。
使用英特尔编译器进行自动向量化 作者：Yang Wang (Intel) 自动向量化是英特尔编译器提供的一个可以自动的使用SIMD指示的功能。在处理数据时，编译器自动选择MMX?, Intel? Streaming SIMD 扩展（Intel? SSE, SSE2, SSE3 和SSE4）等指令集，对数据进行并行的处理。使用编译器提供的自动向量化功能是提高程序性能的一个非常有效的手段。自动向量化在IA-32和Intel? 64的平台上均提供很好的支持。 英特尔编译器提供的自动向量化相关的编译选项如下所示。”/Q”开头的选项是针对Windows平台的，“-“开头的选项是针对Linux*和Mac平台的。 -x, /Qx 按照该选项指定的处理器类型生成相应的优化代码。比如-xSSE3, 该选项指定编译器生成Intel? SSE3指令的代码。又比如-xSSE3_ATOM, 该选项针对Intel? Atom? 处理器进行优化。 -ax, /Qax 如果指定该选项，在生成的单一目标文件中，不但会生成专门针对指定的处理器类型进行优化的代码，同时也生成通用的IA-32架构的代码。该选项主要是为了生成代码的兼容性考虑。 -vec, /Qvec 打开或者关闭编译器的向量化优化。默认情况下自动向量化是打开的。 -vec-report, /Qvec-report 该选项用户控制在编译过程中产生的向量化消息报告。 编译器提供的自动向量化优化默认情况下是打开的。在编译过程中我们可以使用-vec-report选项来打开向量化诊断消息报告。这样编译器可以告诉我们有哪些循环被向量化了，有哪些循环没有被向量化已经无法向量化的原因。 在编译程序的过程中，有时候我们会发现编译器报告说某个循环无法被向量化。很多时候无法向量化的原因都是因为循环中存在的变量依赖关系。有时候我们可以修改程序来消除这种依赖关系，有的时候我们可以使用编译器提供的一些编译指示来显示的告诉编译器如何处理这种依赖关系。即使在某个循环已经可以被自动向量化的时候，使用编译器提供的对向量化的语言支持和编译指示还可以提高编译器向量化的效率，提高程序执行的性能。 下面我们来详细解释一下编译器提供的编译指示以及这些指示对编译器编译的影响。 在Intel编译器中，我们提供下面这样一些对自动向量化的语言支持和编译指示。 __declspec(align(n)) 指导编译器将变量按照n字节对齐 __declspec(align(n,off)) 指导编译器将变量按照n字节再加上off字节的编译量进行对齐 restrict 消除别名分析中的二义性 __assume_aligned(a,n) 当编译器无法获取对齐信息时，假定数组a已经按照n字节对齐 #pragma ivdep 提示编译器忽略可能存在的向量依赖关系 #pragma vector {aligned|unaligned|always}
网络安全课程设计 目录 一、校园网概况 二、校园网安全需求分析 三、产品选型和网络拓扑图介绍 四、操作系统安全配置与测试 五、应用服务器（WWW）安全配置 六、防病毒体系设计 七、防火墙设计、配置与测试 一、校园网概况
该校园网始建于2000年8月，至今已经历了四个主要发展阶段，网络覆盖已遍及现有的教学办公区和学生宿舍区。截止目前，校园网光缆铺设约一万二千米，信息点铺设接近一万，开设上网帐号8000多个，办理学校免费邮箱2000左右。 校园网主干现为双千兆环网结构。校园网接入均为千兆光纤到大楼，百兆交换到桌面，具有良好的网络性能。 校园网现有三条宽带出口并行接入Internet，500兆中国电信、100兆中国网通和100兆中国教育科研网，通过合理的路由策略，为校园网用户提供了良好的出口带宽。 校园网资源建设成效显著，现有资源服务包括大学门户网站、新闻网站、各学院和职能部门网站、安农科技网站、邮件服务、电子校务、毕博辅助教学平台、在线电视、VOD点播、音乐欣赏、公用FTP、文档下载、软件下载、知识园地、站点导航、在线帮助、系统补丁、网络安全、个人主页、计费服务、VPN、DHCP、域名服务等。还有外语学习平台，图书馆丰富的电子图书资源，教务处的学分制教学信息服务网、科技处的科研管理平台等。众多的资源服务构成了校园网的资源子网，为广大师生提供了良好的资源服务。 二、校园网安全需求分析 将安全策略、硬件及软件等方法结合起来，构成一个统一的防御系统，有效阻止非法用户进入网络，减少网络的安全风险。 定期进行漏洞扫描，审计跟踪，及时发现问题，解决问题。 通过入侵检测等方式实现实时安全监控，提供快速响应故障的手段，同时具备很好的安全取证措施。 使网络管理者能够很快重新组织被破坏了的文件或应用。使系统重新恢复到破坏前的状态，最大限度地减少损失。
在双重循环中怎样写OpenMP？ 那要分析你的外循环跟内循环有没有彼此依赖的关系 unsigned int nCore = GetComputeCore(); unsigned int nStep = IMAGETILEYSIZE / nCore; #pragma omp parallel for private(Level0_x, Level0_y, ChangeLevel0_x, ChangeLevel0_y, InterX1, InterX2, InterY1, InterY2) for (int k = 0; k < nCore; k++) { int begin = k * nStep; int end = (k + 1) * nStep; for (int YOff = begin; YOff < end; YOff++) { for (int XOff = 0; XOff < IMAGETILEXSIZE; XOff++) { Level0_x = pPixelXBuf[YOff][XOff]; Level0_y = pPixelYBuf[YOff][XOff]; ChangeLevel0_x = Level0_x - XMin; ChangeLevel0_y = Level0_y - YMin; //寻找坐标在Level1Buf中对应的4个像素值 InterX1 = (int)(ChangeLevel0_x); InterX2 = (int)(ChangeLevel0_x + 1); InterY1 = (int)(ChangeLevel0_y); InterY2 = (int)(ChangeLevel0_y + 1); //双线性插值对Level0_Buf赋值 ZoomInterpolation(Level0Buf, Level1Buf, ChangeLevel0_x, ChangeLevel0_y, SamplesPerPixel, nXSize, nYSize, InterX1, InterX2, InterY1, InterY2, XOff, YOff); } } } 我也想应该这样，可是如果nCore=1的时候，外循环只循环一次，线程是怎么分配的呢。其实最外层的循环如果很多，就在外循环分配线程是不是就可以不考虑里面的循环了？ nCore = 1,就是单核单cpu,多核循环就跟普通的循环没有差别， openmp默认有几个内核就开几个线程同时运行。所以单核openmp也没有什么意义，此时你也可以开两个线程“同时”运行，但在单核机器上，两个线程是不可能同时运行的 可以不考虑里面的循环。你只要保证外循环跟外循环之间，内寻环跟内循环之间没有数据依赖关系就行。 假设 for (int i = 0; i < 200000000; i++)
课程设计任务书 题目：校园网络安全解决方案 学号： 姓名： 专业： 课程： 指导老师：职称： 完成时间：2007年 12 月----2008年 1月**学院计算机科学系制
校园网络安全解决方案 引言：校园网网络是一个分层次的拓扑结构，因此网络的安全防护也需采用分层次的拓扑防护措施。即一个完整的校园网网络信息安全解决方案应该覆盖网络的各个层次，并且与安全管理相结合。 一、网络信息安全系统设计原则 1.满足Internet分级管理需求 2.需求、风险、代价平衡的原则 3.综合性、整体性原则 4.可用性原则 5.分步实施原则 目前，对于新建网络及已投入运行的网络，必须尽快解决网络的安全保密问题，设计时应遵循如下思想： 大幅度地提高系统的安全性和保密性； 保持网络原有的性能特点，即对网络的协议和传输具有很好的透明性； 易于操作、维护，并便于自动化管理，而不增加或少增加附加操作； 尽量不影响原网络拓扑结构，便于系统及系统功能的扩展； 安全保密系统具有较好的性能价格比，一次性投资，可以长期使用； 安全与密码产品具有合法性，并便于安全管理单位与密码管理单位的检查与监督。 基于上述思想，网络信息安全系统应遵循如下设计原则： 满足因特网的分级管理需求根据Internet网络规模大、用户众多的特点，对Internet/Intranet信息安全实施分级管理的解决方案，将对它的控制点分为三级实施安全管理。 第一级：中心级网络，主要实现内外网隔离；内外网用户的访问控制；内部网的监控；内部网传输数据的备份与稽查。 第二级：部门级，主要实现内部网与外部网用户的访问控制；同级部门间的访问控制；部门网内部的安全审计。 第三级：终端/个人用户级，实现部门网内部主机的访问控制；数据库及终端信息资源的安全保护。 需求、风险、代价平衡的原则对任一网络，绝对安全难以达到，也不一定是必要的。对一个网络进行实际额研究(包括任务、性能、结构、可靠性、可维护性等)，并对网络面临的威胁及可能承担的风险进行定性与定量相结合的分析，然后制定规范和措施，确定本系统的安全策略。 综合性、整体性原则应用系统工程的观点、方法，分析网络的安全及具体措施。安全措施主要包括：行政法律手段、各种管理制度(人员审查、工作流程、维护保障制度等)以及专业措施(识别技术、存取控制、密码、低辐射、容错、防病毒、采用高安全产品等)。一个较好的安全措施往往是多种方法适当综合的应用结果。一个计算机网络，包括个人、设备、软件、数据等。这些环节在网络中的地位和影响作用，也只有从系统综合整体的角度去看待、分析，才能取得有效、可行的措施。即计算机网络安全应遵循整体安全性原则，根据规定的安全策略制定出合理的网络安全体系结构。 可用性原则安全措施需要人为去完成，如果措施过于复杂，要求过高，本身就降低了安全性，如密钥管理就有类似的问题。其次，措施的采用不能影响系统
sizeof用法总结 本文主要包括二个部分，第一部分重点介绍在VC中，怎么样采用sizeof来求结构的大小，以及容易出现的问题，并给出解决问题的方法，第二部分总结出VC中sizeof的主要用法。 1、sizeof应用在结构上的情况 请看下面的结构： struct MyStruct { double dda1; char dda; int type }; 对结构MyStruct采用sizeof会出现什么结果呢？sizeof(MyStruct)为多少呢？也许你会这样求： sizeof(MyStruct)=sizeof(double)+sizeof(char)+sizeof(int)=13 但是当在VC中测试上面结构的大小时，你会发现sizeof(MyStruct)为16。你知道为什么在VC中会得出这样一个结果吗？ 其实，这是VC对变量存储的一个特殊处理。为了提高CPU的存储速度，VC对一些变量的起始地址做了“对齐”处理。在默认情况下，VC规定各成员变量存放的起始地址相对于结构的起始地址的偏移量必须为该变量的类型所占用的字节数的倍数。下面列出常用类型的对齐方式(vc6.0,32位系统)。 类型 对齐方式（变量存放的起始地址相对于结构的起始地址的偏移量） Char 偏移量必须为sizeof(char)即1的倍数 int 偏移量必须为sizeof(int)即4的倍数 float 偏移量必须为sizeof(float)即4的倍数 double 偏移量必须为sizeof(double)即8的倍数
Short 偏移量必须为sizeof(short)即2的倍数 各成员变量在存放的时候根据在结构中出现的顺序依次申请空间，同时按照上面的对齐方式调整位置，空缺的字节VC会自动填充。同时VC为了确保结构的大小为结构的字节边界数（即该结构中占用最大空间的类型所占用的字节数）的倍数，所以在为最后一个成员变量申请空间后，还会根据需要自动填充空缺的字节。 下面用前面的例子来说明VC到底怎么样来存放结构的。 struct MyStruct { double dda1; char dda; int type }； 为上面的结构分配空间的时候，VC根据成员变量出现的顺序和对齐方式，先为第一个成员dda1分配空间，其起始地址跟结构的起始地址相同（刚好偏移量0刚好为sizeof(double)的倍数），该成员变量占用sizeof(double)=8个字节；接下来为第二个成员dda分配空间，这时下一个可以分配的地址对于结构的起始地址的偏移量为8，是sizeof(char)的倍数，所以把dda存放在偏移量为8的地方满足对齐方式，该成员变量占用sizeof(char)=1个字节；接下来为第三个成员type分配空间，这时下一个可以分配的地址对于结构的起始地址的偏移量为9，不是sizeof(int)=4的倍数，为了满足对齐方式对偏移量的约束问题，VC自动填充3个字节（这三个字节没有放什么东西），这时下一个可以分配的地址对于结构的起始地址的偏移量为12，刚好是sizeof(int)=4的倍数，所以把type存放在偏移量为12的地方，该成员变量占用sizeof(int)=4个字节；这时整个结构的成员变量已经都分配了空间，总的占用的空间大小为：8+1+3+4=16，刚好为结构的字节边界数（即结构中占用最大空间的类型所占用的字节数sizeof(double)=8）的倍数，所以没有空缺的字节需要填充。所以整个结构的大小为： sizeof(MyStruct)=8+1+3+4=16，其中有3个字节是VC自动填充的，没有放任何有意义的东西。 下面再举个例子，交换一下上面的MyStruct的成员变量的位置，使它变成下面的情况： struct MyStruct { char dda; double dda1;