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Long-distance travel to provide mental health services for deaf people has implications for efficiency, safety, and equality of service. However, uptake of Telemental Health (TMH) has been slow in both deaf and general mental health services. A quantitative study was used to investigate access to TMH and whether staff confidence, experience, or demographics affect TMH use. It was concluded that staff in neither deaf mental health services nor general mental health services had adequate knowledge of or access to TMH. Staff expressed concerns over TMH's appropriateness in their work. Previous use of videoconferencing was assosciated significantly with confidence, but previous use of videophones was not. Neither staff in deaf services nor deaf staff were more experienced with or more confident about videoconferencing, whereas, within deaf services, deaf staff were significantly more confident about videophone use. Training implications are discussed.

We propose a computational opinion model for estimating video quality of videophone services. Opinion models for speech such as the E-model have been established and widely used; however, little attention has been given to opinion models for video quality estimation. Our proposed opinion model is useful as a network-planning tool for assessing several video parameters that affect the quality of videophone services. First, we established a function for estimating video quality affected by coding degradation, which expresses the quality of a video affected by coding bit rate and frame rate. Second, we established a packet loss degradation index that estimates the degree of video quality degradation due to packet loss. Finally, we integrated these two functions into the opinion model for estimating video quality. We applied this model for video quality estimation of a videophone service with various video formats and displayed video sizes. The results indicated that the estimation errors of our model were less than the mean of the 99% confidence intervals for the subjective scores. Therefore, our model could be applied to effective design, implementation, and management of videophone applications and communication networks.

Reminiscence therapy, which is effective for increasing the selfesteem of and for reducing behavioral disturbances in individuals with dementia, is usually conducted in a group led by experienced staff. However, due to the shortage of care attendants, only a limited number of patients at home can receive the benefits of this therapy. To provide this therapy for patients anytime or anywhere, we have developed a networked reminiscence therapy system that combines IP videophones with a photo- and video-sharing mechanism based on Web technology. First, we prepared the experimental setup in a hospital and examined whether dementia patients could communicate with therapists by videophone. Then we conducted a field trial of networked reminiscence therapy with a more realistic situation where remote volunteers communicated with dementia sufferers in the care home by IP videophones connected by broadband network. In this paper, we describe our developed system. Then, we present experimental results showing that dementia sufferers could communicate with therapists by videophone and that networked reminiscence sessions were generally as successful for individuals with dementia as face-to-face reminiscence sessions.

In this age of multimedia convergence, video based applications for conferencing and streaming are seeing big market traction. The main challenge of implementing these systems is a tradeoff between several factors like bandwidth, image quality, implementation cost and speed (in terms of mega cycles per second). Moreover to implement this encoder-decoder system (CODECs) in a real time system based on any digital signal processor (DSP) is more difficult because of their restricted resources like memory and CPU speed. In this paper, we present some novel approach, which reduces the computational complexity and also meets the memory constraint of the target processor for a standard H.264 baseline encoder without sacrificing the rate-distortion performance. The proposed algorithms are applied on standard test sequence of different resolutions. The results obtained from a considerably large number of test sequences show the strength of our proposed algorithm. We claim the betterment in performance measured in mega cycles per second (MCPS) at the cost of negligible loss in image quality. Moreover we gain the betterment in image quality using algorithm modification in rate controlling. Implementation details are also presented for a QCIF H.264 baseline encoder in a TMS320C55x DSP, which has only 256 KB RAM and 150 MHz clock speed. The enhanced performance is achieved thorough formulation of novel adaptive algorithms for rate control and motion estimation. We achieve almost 40% MCPS reduction at the cost of less than 1% reduction in image quality. We also get betterment in image quality by our proposed rate-controlling algorithm

A wavelet-based videophone scheme proposed, where the video bits are unequal error protection (UEP) using irregular convolutional codes (IRCCs). The proposed system uses adaptive arithmetic coding (AAC) for encoding the motion vectors and individual wavelet subband coefficients. The turbo-equalized IRCC-aided videophone scheme is capable of attaining a near unimpaired video quality for channel signal-to-noise ratios (SNRs) in excess of about 4.5 dB over a five-path dispersive AWGN channel

The videophones based on Internet protocol (IP) are gaining increasing acceptance because of the enhanced compression capabilities using H.264/MPEG-4 AVC. H.264 offers high compression, but it also demands high degree of complexity. Hence, optimization to the maximum extent is required for designing the H.264 video encoder with real time performance on DSPs. In this paper a highly optimized H.264 baseline profile video encoder on a programmable DSP processor is presented.

In this paper, we present an efficient face transmission system for MPEG-4 mobile videophone via the wireless Internet. The system consists of two stages: region segmentation and facial region detection. In the region segmentation, the input image is segmented into an appropriate set of arbitrary regions using a watershed algorithm. In facial region detection, the facial regions are identified from the segmented results using a skin-color model. The results of region segmentation and facial region detection are integrated to provide facial regions with accurate and closed boundaries. This system can be used to transmit a face to a corresponding person via the wireless Internet.

In The case of a system for delivering “live streaming” video and audio from the internet under a videophone mechanism in real-time, the major issue is to assure video quality and usability at the same time. To solve this issue, NTT DoCoMo has developed technology that utilizes the “video-streaming gateway” provided by “M-stage V Live” service.

We propose a Vector-Quantized (VQ) video codec that induces deliberately-introduced code constraints. The complete set of these code constraints are unambiguously described by a novel block-based modification of a symbol-based variable length coding trellis structure. This trellis is employed during VQ encoding to represent video information with a Minimum Mean Squared Error (MMSE) reconstruction. Additionally, the proposed trellis structure is employed in the video decoder to obtain a posteriori probability based MMSE estimates of the transmitted VQ tiles. Since the trellis structure unambiguously describes the code constraints induced by the video codec, its employment guarantees the automatic recovery of valid sets of VQ tiles. The proposed video codec is serially concatenated with a trellis-coded modulation scheme for transmission over a fast-fading Rayleigh fading channel. In the receiver, iterative-decoding convergence that eliminates all perceivable channel-induced video distortion is shown to be supported within 1.29 dB of the system's 2 bit/s/Hz channel-capacity limit of 3.96 dB.

With the development of the broadband network, video communications such as videophone, video distribution, and IPTV services are beginning to become common. In order to provide these services appropriately, we must manage them based on subjective video quality, in addition to designing a network system based on it. Currently, subjective quality assessment is the main method used to quantify video quality. However, it is time-consuming and expensive. Therefore, we need an objective quality assessment technology that can estimate video quality from video characteristics effectively. Video degradation can be categorized into two types: spatial and temporal. Objective quality assessment methods for spatial degradation have been studied extensively, but methods for temporal degradation have hardly been examined even though it occurs frequently due to network degradation and has a large impact on subjective quality. In this paper, we propose an objective quality assessment method for temporal degradation. Our approach is to aggregate multiple freeze distortions into an equivalent freeze distortion and then derive the objective video quality from the equivalent freeze distortion. Specifically, our method considers the total length of all freeze distortions in a video sequence as the length of the equivalent single freeze distortion. In addition, we propose a method using the perceptual characteristics of short freeze distortions. We verified that our method can estimate the objective video quality well within the deviation of subjective video quality.

The chapter reviews two main video call protocols—H.323 and SIP—and the video and audio codecs that are used in video calls. It explains how to select a video call software solution, what is the criteria to rate the solution in the market, who are the various video telephone manufacturers, and where to find more about their solutions because many do not have retail offerings. There are more than 40 solutions of video call software in the market. The chapter highlights various video telephone devices that the VoIP services offer. Primarily, there are two options for video telephones—ones that require a VoIP service provider, such as Packet-8, Vonage, and ACN, and the ones that do not, such as D-Link, Black Crow, and the dial-up solution from Vialta. The chapter helps customers understand the options available for video telephones, and how to start deciding what is suitable and what is not. The chapter also explains where to find long RCA cables to connect a video telephone to a television.

High speed connections have enabled video-based communication and given deaf and hard-of-hearing people the opportunity to rely on the visual modality to better communicate with each other and with hearing people. However a number of obstacles still have to be removed to make this technology really accessible for all: technical expertise for operating a computer, lack of compatibility, poorly integrated services,... This papers describes an ongoing project for providing a better accessibility by (1) supporting multiple kind of user terminals (such as PC, videophone, TV, mobile phone) in an interoperable way; (2) with a simple and user friendly interface, (3) giving access to a number of related services such as directories, video message box, relay centre and signed news

Imaging performance and color rendering property of a free-form optical prism that plays camera lens and projection display are investigated. Complementary designs in photoptic and scotopic environments are developed for superior color adaptation and optimization.

Most multimedia source signals are capable of tolerating lossy, rather than lossless delivery to the human eye, ear and other human sensors. The corresponding lossy and preferably low-delay multimedia source codecs however exhibit unequal error sensitivity, which is not the case for Shannon’s ideal entropy codec. This paper proposes a jointly optimised turbo transceiver design capable of providing unequal error protection for MPEG-4 coding aided wireless video telephony. The transceiver investigated consists of space-time trellis coding (STTC) invoked for the sake of mitigating the effects of fading, in addition to bandwidth efficient trellis coded modulation or bit-interleaved coded modulation, combined with a multi-level coding scheme employing either two different-rate non-systematic convolutional codes (NSCs) or two recursive systematic convolutional codes for yielding a twin-class unequal-protection. A single-class protection based benchmark scheme combining STTC and NSC is used for comparison with the unequal-protection scheme advocated. The video performance of the various schemes is evaluated when communicating over uncorrelated Rayleigh fading channels. It was found that the proposed scheme requires about 2.8 dBs lower transmit power than the benchmark scheme in the context of the MPEG-4 videophone transceiver at a similar decoding complexity.

This paper presents an architecture for a multimedia application platform able to support real-time group communication activities (such as: Videophone, Teletraining, Computer Supported Collaborative Work) between remotely located members of groups by delivering multimedia communication capabilities directly at their personal computers ("desktops"). The paper focuses on a component-based design approach that gives the potential to integrate into the platform various Application Components (Audio, Video, Text chat, File transfer, Slide show, Shared Whiteboard or text editor, etc.) depending on the activity supported and thus permitting to customise the platform for the specilic activity. Apart from the basic architecture, alternative connection schemes depending on the capabilities of the network are surveyed. A discussion of the advantages and disadvantages of the described architecture is also given andfinally, an-implernentation aiming to support Teletraining activities is presented.

An economical multipoint audiovisual communicating system technology for the ISDN basic rate interface is introduced. The system offers multipoint audiovisual communication using the ring-type connection via the two B channels in the ISDN basic rate interface for each terminal. The system's connection control offers a terminal skipping procedure. Multipoint image processing produces a divided display by digital data manipulation. Finally, an ISDN basic rate interface videophone using this system technology is described.

Video-mediated communication systems such as teleconferencing and videophone have become popular. As with face-to-face communication, non-verbal cues such as gaze, facial expression, head orientation and gestures in visual systems play an important role. Existing systems, however, do not support mutual gaze because the lay-out of the camera and monitor is restricted. Thus, conversations using visual systems differ from those in face-to-face communication. This paper clarifies the problems of the video-mediated system, specifically for comparing the system with communication using eye-contact and with communication using no-eye-contact. This study focuses on the protocol of opening communication, e.g. establishment of a visual-audio link, person identification and confirmation of the acceptance of conversation. We conducted experiments using the two systems. Analysis of recorded video sequences revealed that the system using communication with eye-contact induced behavior similar to the system using face-to-face communication.

Architecture of an IP-based narrow-band videophone system is proposed in this paper for convenient videophone calls between any two computers even if being placed in two different LANs within network agents. The bandwidth need of each call is less than 256 kbps. The system consists of two kinds of entities: Videophone Terminals (VPT) and a Video Call Server (VCS). A VPT is actually a microcomputer program, composed of 4 primary parts, an audio codec, a video codec, a media deliverer/receiver and a call controller. The basic functions of the VCS include videophone number generation and management, access admission and address resolution. The VCS with a public IP address plays an important role in the system especially when a video call has to penetrate through network agents. Each VPT in the system gets its own external transport address from the VCS through registration process. A calling VPT would receive the external transport address of the called VPT from the VCS through address resolution. The proposed system works and is helpful to accelerate the realization of people's videophone dream over IP networks.

Mobile phones have become essential part of modern life and continue to change the way people communicate with each other. The camera phones have become ubiquitous in the recent past and video communication services such as videophone are getting rolled out by cellular providers. This would require video compression algorithms that are resilient to transmission errors and support resource-adaptive playback. We propose a video codec that addresses the two key issues in video delivery to mobile devices: (1) error resilience; and (2) resource adaptive playback. We show that the proposed codec is simple to implement on low-resource terminals such as mobile phones and outperforms MPEG-4 coding in terms of loss tolerance.

We have designed and tested an Internet-based video-phone suitable for use in the homes of families in need of paediatric palliative care services. The equipment uses an ordinary telephone line and includes a PC, Web camera and modem housed in a custom-made box. In initial field testing, six clinical consultations were conducted in a one-month trial of the videophone with a family in receipt of palliative care services who were living in the outer suburbs of Brisbane. Problems with variability in call quality—namely audio and video freezing, and audio break-up — prompted further laboratory testing. We completed a programme of over 250 test calls. Fixing modem connection parameters to use the V.34 modulation protocol at a set bandwidth of 24kbit/s improved connection stability and the reliability of the video-phone. In subsequent field testing 47 of 50 calls (94%) connected without problems. The freezes that did occur were brief (with greatly reduced packet loss) and had little effect on the ability to communicate, unlike the problems arising in the home testing. The low-bandwidth Internet-based video-phone we have developed provides a feasible means of doing telemedicine in the home.

A perceptual weighting model is proposed for effective rate control so as to enhance the perceptual coding quality of videophone. We exploit two categories of factors affecting the perception of the human visual system, stimulus-driven factors and cognition-driven factors. In order to achieve a simple, but effective, perceptual weighting model, we use luminance adaptation and texture masking as the stimulus-driven factors, while skin color serves as the cognition-driven factor in the videophone application. Both objective and subjective quality evaluations of videophone-like sequences in the H.263 platform validate the effectiveness of our perceptual weighting model.

We present a method for extracting local visual perceptual clues and its application for rate control of videophone, in order to ensure the scarce bits to be assigned for maximum perceptual coding quality. The optimum quantization step is determined with the rate-distortion model considering the local perceptual clues in the visual signal. For extraction of the perceptual clues, luminance adaptation and texture masking are used as the stimulus-driven factors, while skin color serves as the cognition-driven factor in current implementation. Both objective and subjective quality evaluations are given by evaluating the proposed perceptual rate control (PRC) scheme in the H.263 platform, and the evaluations show that the proposed PRC scheme achieves significant quality improvement in the block-based coding for bandwidth-hungry applications.

This paper focuses on the interactions between uplink admission and congestion control strategies in UTRA-W-CDMA. The proposed strategies are analysed in the framework of a representative multiservice scenario where conversational and interactive traffic are present and the presented solutions are compliant with 3GPP UTRA FDD specifications. Furthermore, the results consider a complete approach where all the radio resource management strategies are taken into account, including admission, congestion, short term RRM, handover and power control feasible solutions. Therefore, obtained results allow to devise a range of guidelines for the joint design of RRM algorithms.

In this paper, the authors reported the use of an e-learning system based on a mobile videophone in an education program at home for a student with severe physical impairments. System use for two months showed that the subject was able to attend school curricula and engage in discussion with teachers and classmates through the system.

This article uses a human factors evaluation framework to assess the usability of commercially available videophone technology. One study focuses on minimally functionally impaired seniors living in an assisted living facility. The second study focuses on usability for hospice staff. Seniors found the technology easy to use and were willing to accept the equipment in their homes, especially if requested by a healthcare provider. Administrators and hospice care providers also reported that the videophones were easy to use and would be of benefit to the patients they care for. The results indicate that videophones are a promising intervention with identifiable limitations.

Moving object extraction plays a key role in applications such as object-based videoconference, surveillance, and so on. The difficulties of moving object segmentation lie in the fact that physical objects are normally not homogeneous with respect to low-level features and it's usually tough to segment them accurately and efficiently. Object segmentation based on prestored background information has proved to be effective and efficient in several applications such as videophone, video conferencing, and surveillance, etc. The previous works, however, were mainly concentrated on object segmentation with a static camera and in a stationary background. In this paper, we propose a robust and fast segmentation algorithm and a reliable tracking strategy without knowing the shape of the object in advance. The proposed system can real-time extract the foreground from the background and track the moving object with an active (pan-tilt) camera such that the moving object always stays around the center of images.

This chapter focuses on screen and video phones. Screenphones are the corded cousins of smart phones. Screenphones offer many of the same options as smart phones, including Internet access, shopping, and email. However, screenphones do have one important advantage over smart phones—the viewable screen area can be significantly larger than that of a smart phone, thus making Web browsing easier on the eyes. Screenphone commands and functions are activated on the virtual keyboard by touching desired keys and icons with a finger or specially designed pen. The screenphone communicates cordlessly via Bluetooth with the compact, color screen that can be carried from room-to-room. It provides Internet access for applications such as email, messaging, informational services, and Web browsing. The screenphone delivers features driven by concept familiarity, product suitability, and existing network/service support. Screenphones also include several useful applications such as broadband transmission and reception, email, Internet connectivity, safe and secure transactions, PC-like features, and Internet phone calls that allow the user to call and surf at the same time. Furthermore, another popular category of Internet videophones has hit the market. These phones incorporate an Internet videophone, the PC, digital camera, microphone, speakers, and a modem. These videophones use the Internet/IP backbone to transmit voice, video, and data among users based miles apart.

A jointly optimised turbo transceiver capable of providing unequal error protection is proposed for MPEG-4 aided wireless video telephony. The transceiver advocated consists of Space-Time Trellis Coding (STTC), Trellis Coded Modulation (TCM) and two different-rate Non-Systematic Convolutional codes (NSCs). A benchmarker scheme combining STTC and NSC is used for comparison with the proposed scheme. The video performance of the both schemes is evaluated when communicating over uncorrelated Rayleigh fading channels. It was found that the proposed scheme requires about two dBs lower transmit power than the benchmarker scheme in the context of the MPEG-4 videophone transceiver, when aiming for an effective throughput of 2 bits/symbol at a similar decoding complexity.

We propose an intelligent easy-operating human-machine interface incorporating face detection technology for a mobile multimedia terminal. We have developed a face detection algorithm which works on the resources of a mobile phone. This algorithm is robust in outside fields, and simultaneously works with MPEG-4 Simple Profile Level 1 encoding and decoding on an MPEG-4 codec LSI for a mobile videophone. This enables more intelligent interfaces in visual communication such as IMT-2000 videophone services.

ATM network impairments in terms of CER, CLR, CMR, CDV and CTD are applied to an experimental 3D videophone for audio and 3D video signals quality evaluation study. The effects of the impairments on both signals are evaluated by comparison between unimpaired and impaired sequences. A subjective assessment method together with the IT-R five-grade impairment scale is adopted in performing the evaluation. The results show ATM network impairments effects on both signals can maintain scale 4 of the IT-R five-grade scale. Therefore, good quality of audio and 3D video signals transmission can be confirmed for IT-T ATM performance objectives.

This paper proposes a new concept of mobile telephone with a PC resolution virtual display that has the following five functions: telephone, E-mail, Web telephone, videophone and VOD (video on demand) viewer. We developed a concept proof prototype based on this concept.

A new method is proposed to automatically segment out a person's face from a given sequence of images that consists of a head-and-shoulder view, using the fast marching level set approach. The method proposed involves a fast, reliable and computationally efficient algorithm, which exploits the colour information in an image to segment the face region. The colour information is derived and used to classify pixels in the image into different levels depending on the value of the colour information for that pixel. Once this is achieved, a priority-labelled fast marching level set algorithm is utilised to locate the face region. The performance of the fast marching face-segmentation algorithm is illustrated by some simulation results carried out on head-and-shoulder test images.

This paper presents a performance study on MPEG-4 coder in videotelephony applications. This study addresses both MPEG-4 frame-based and object-based approaches to encoding videophone image sequences. The overhead cost, bit/quality transfer capability among video objects, and the use of video objects that are defined at different image resolutions are investigated.

To overcome the bandwidth limitation and to strengthen the channel effect immunity, a cube-based polynomial video coding scheme for mobile radio transmission is derived, named three-dimensional polynomial approximation coding (3D PAC). It also possesses the features of low bit rate of transmission data, robustness for channel fading and simple computation scheme for real-time processing. Compared with the videophone standard H.261, it gains advantage on resisting channel noise to 2 dB with same bit rate, and the superior subjective performance may be obtained.

An image coding algorithm that enables hand sign language to be used for conversation between the deaf over a telephone network is described. A so-called valledge detector produces cartoonlike images at a resolution of 64 pels by 64 pels. The images are compressed using a block oriented algorithm and transmitted at 9.6 kb/s using full duplex modems. TMS32010 digital signal processors are use to implement the coding algorithm in real time. Preliminary tests using 14 profoundly deaf hand signers were very positive, indicating that the valledge-detected pictures are usable for fairly normal hand signing conversations, with only mildly impaired efficiency compared to full-bandwidth video pictures. >

The authors describe an approach to the coding of moving sequences that combines image segmentation with adaptive thresholding based on a priori knowledge about the scene and motion-adaptive frame interpolation techniques. In videoconference pictures the face is an area of high movement, and it is also the most noticeable part of the picture. Using such knowledge about the scene, the location of the face in alternate frames is determined, and a set of thresholds is chosen that allows segmentation of this part of the image more accurately than of the background. To achieve the high compression ratios required for transmission of videoconference and videophone pictures over the existing networks, only alternate frames are coded and transmitted. At the receiver a skipped frame k is produced by motion-adaptive interpolation of corresponding blocks of frames k-1 and k+1. >

A low-bit-rate image codec for videophone and B/2B-ISDN (integrated services digital network) applications is presented. The encoder skips two frames over three images and encodes one image over three, using DCT (discrete cosine transform)-Huffman coding, and sends this information together with the motion vectors computed on overlapped blocks. At the receiver the missing frames are reconstructed by linear interpolation using coded frames and corresponding motion vectors. The resulting video sequences indicate that for a videophone in ISDN network applications, an acceptable quality image is obtained. >

ISBN 0818619716; Special-purpose chips based on highly parallel architectures, for possible use in videophone, videoconference, digital TV distribution and HDTV codecs, are considered. Functional interface specifications for a motion estimator to be inserted in a coder system are detailed. The range of possible operative part-kernel architectures is derived, based on formal high-level synthesis methods. A novel implementation using redundant base 2 digital-serial arithmetic is presented, resulting in a compact (70000 transistors) motion estimation VLSI block.

A method is described which uses a form of subband coding which is efficient and simple, satisfies some of the most likely constraints of packet networks, and exploits some of their features. Results are reported on the application of the method to broadcast-quality moving pictures and to videoconference or videophone pictures. Simulation results have shown the method to be very attractive both for high bit rate (a few tens of megabits per second) and low bit rate (some hundreds of kilobits per second) in terms of performance. The method has some features that recommend it in case interworking between different networks is required (e.g. between LANs and ISDN), and it is well matched to the current standardization plans of the CCITT. >

A description is given of studies of source coding in the specialists group on coding for visual telephony in CCITT SG XV. Using a macro block approach, the source coding algorithm will have the capability of operating from 64 kb/s for videophone applications up to 2 Mb/s for high-quality videoconferencing applications. The authors describe the existing reference model and the evolution toward the latest reference model (RM5), which is going to be submitted for standardization. Theoretical information is provided and some examples for possible improvements are presented for the most important techniques used in the reference model, among which are quantization, variable block size, scanning classes, loop filter, entropy coding, multiple vs. single variable-length coding, and block-type discrimination. >

An innovative knowledge-based scheme is presented for videophone sequence coding, with a reasonable subjective reconstruction quality. The videophone images are segmented and approximated by means of a suitably adapted 3D wire-frame model capable of faithfully reproducing the signal temporal evolution through a limited number of updating parameters extracted from the real sequence. The muscle modeling represents the basic issue for the analysis of facial expressions at the encoder and for their synthesis at the decoder. Facial muscles are modeled either as linear fibers or as circular fibers and are characterized by descriptive parameters such as thickness, length, position and direction. The animation is performed by simulating the muscle contraction and relaxation over the domain of influence defined by the muscle parameters. The interactive procedure for the construction and adaptation of the 3D model and the analysis/synthesis algorithms, implemented on a multiprocessor AT&T Pixel Machine, are described, together with some examples of the achieved experimental results. >

The authors describe the design of a video codec CCITT H.261 on 2 PC/AT boards based on special purpose DSPs, RAMs and Xilinx LCAs. The CCITT H.261 recommendations describes the video coding and decoding methods for the moving picture component of audiovisual services at the rates of p*64 kbit/s, where p is in the range 1 to 30. The codec has been developed jointly with Centre National d'Etudes des Telecommunications and is now connected to ISDN for videophone and videoconferencing communications. In addition to describing the architecture and design, the authors discuss the development system used in the implementation. >

The design and implementation of the user-network signaling protocol and call processing software in a desktop multimedia conferencing system are described. The system is built on an EXPANSE broadband network prototype and aims at supporting multimedia and multiparty services such as videophone, video mail, video-on-demand, and video conferencing. A message-based, common-channel signaling protocol is used. It supports multiple connections and multiple parties within a call. The call processing software has a layered structure and is implemented in C++ using an object-oriented approach. The inheritance and virtual function mechanisms of C++ also ease the introduction of new services to the system. >

Research on high-capacity mobile communications is approaching modems capable of transmitting 64 kb/s in a 30-kHz channel. This rate has made it possible to conceive of new mobile services, the most ambitious of which is perhaps video. Although data rates of 64 kb/s are thought of as very high in the mobile environment, this is a very low rate for video. Many techniques for implementing video at rates between 64 and 128 kb/s have focused on teleconferencing or videophone over a high-quality channel. The results of the investigation into how to adapt the p*64 video coding standard so that it will be suitable for the coding of QCIF formal (176*144) video sequences for mobile transmission are presented. It is concluded that the codec design presented should be robust enough to allow reasonable quality video transmission over a mobile channel operating at 64 kb/s. >

A stand-in is a common technique for movies and TV programs in foreign languages. The current stand-in that only substitutes the voice channel results awkward matching to the mouth motion. Videophone with automatic voice translation are expected to be widely used in the near future, which may face the same problem without lip-synchronized speaking face image translation. In this paper, we propose a method to track motion of the face from the video image and then replace the face part or only mouth part with synthesized one which is synchronized with synthetic voice or spoken voice. This is one of the key technologies not only for speaking image translation and communication system, but also for an interactive entertainment system. Finally, an interactive movie system is introduced as an application of entertainment system.

The H.261 standard defines the video coding and decoding methods for digital trans mission over ISDN at rates of p x 64kbit/s, where p is in the range of 1-30. The video bit rates will lie between approximately 64kbit/s and 1920kbit/s. The recommendation is aimed at meeting projected customer demand for videophone, videoconferencing and other audio-visual services. The coding structure of H.261 is very similar to that of the generic codec of Chapter 3. That is, it is an interframe DCT-based coding technique. Interframe prediction is first carried out in the pixel domain. The prediction error is then transformed into the frequency domain, where the quantisation for bandwidth reduction takes place. Motion compensation can be included in the prediction stage, although it is optional. Thus the coding technique removes temporal redundancy by interframe prediction and spatial redundancy by transform coding. Techniques have been devised to make the codec more efficient, and at the same time suitable for telecommunications.

For scalable and model-based coding of a videophone sequence at very low bit rates, a 2D scalable face model has to be designed. An efficient algorithm is presented for the automatic detection of a chin contour in a human face. The chin is first represented by a deformable template consisting of two parabolas. Then, a cost function is minimized to find the best fit of the template to the chin. Finally, the chin contour is detected using the active snake algorithm, which is initialised by the best fit of the template. In order to obtain the smoother external force of the active snake model, gradient vector flow (GVF) is used, which is derived from the edge distribution. Experimental results show that the proposed method can drive the snake model to the most optimal chin position. This method can be used for 2D scalable face model design and 3D face model adaptation.