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Most of consumer electronic devices, including HDTV/IPTV, multimedia teleconference systems, on-line game stations, and HiFi-audio systems, are expected to be connected through WiFi WLANs in home/office networks. Especially, the realtime multimedia applications are requiring guaranteed bandwidth to provide guaranteed QoS/QoE. As a consequence, correct estimation of the available bandwidth in overlapped WiFi WLANs in home/office network environments is one of the essential functions for efficient provisioning of Qos-guaranteed realtime multimedia applications and optimized network resource utilizations in future internet. in this paper, we propose a cognitive passive estimation of the available bandwidth (cPEAB) by correct measurements of i) the proportion of waiting and backoff delay, ii) packet collision and packet error probability, iii) acknowledgement delay, iv) channel idle time compared to measurement period, and v) traffic from hidden/exposed terminals. Using a series of experiments, we found that the proposed scheme estimates the available bandwidth more accurately compared with other schemes. [ABSTRACT FROM AUTHOR]

The emergence of three dimensional (3D) video applications, based on Depth Image Based Rendering (DIBR) has brought up more requirements of bandwidth, due to the need of depth information. This additional bandwidth requirement need to be tackled to enable the widespread of 3D video applications based on DIBR. Exploiting visual correlations between the color image and the depth image, in depth image coding, will reduce the requirement of high bandwidth required to transmit the additional depth information. In this paper, an object based depth image coding technique is presented which is suitable for low bit rate 3D-TV applications that are based on Depth Image Based Rendering. The proposed method achieves at most 50% bit rate reduction at low bit rates. [ABSTRACT FROM AUTHOR]

An efficient mobile video streaming system needs to cope with unstable network bandwidth and limited battery life. We propose a novel streaming system which jointly considers picture quality, bit-rate and energy consumption. Reducing the spatial resolution of a video stream adaptively proves to be more efficient in terms of picture quality and energy consumption than conventional rate control using only adjustment of quantization parameter. We apply the same scheme to scalable coding for large-scale mobile video streaming. This extends the adaptation of an SVC stream to lower bit-rates, while maintaining temporal stability. Our approach has been shown to improve picture quality by approximately 0.5dB in low bit-rate conditions, and also reduces energy consumption by more than 50% compared to conventional video streaming. [ABSTRACT FROM AUTHOR]

Artificial bandwidth extension methods aim to improve the quality and intelligibility of narrow band telephone speech by adding new, artificially generated spectral content to the highband of the received voice signal. The development cycle of an artificial bandwidth extension method from the initial idea to the implementation in a mobile terminal is discussed in this paper. Developing the algorithm in the Matlab environment was the first step in the process. The method was then evaluated informal listening tests and simulations to verify its performance in different scenarios. Finally, the utilization of this technology in a product included its DSP implementation combined with the acoustical design of the user terminal. [ABSTRACT FROM AUTHOR]

Streaming high quality audio/video (A V) from home media sources to TV sets and audio speakers over a wireless local area network (WLAN) is a challenging problem because of the fluctuating bandwidth caused by interference and fading. Retransmission and data buffering, as common techniques in data transfer over error-prone channels, are not adequate for delay sensitive applications such as high quality audio streaming. Our approach is to adjust the audio data rate dynamically in order to improve the perceptual quality of audio according to wireless channel quality. In this paper we specifically focus on robust audio streaming over short range wireless channels. The major strength of the proposed algorithm is in its ability to match the source and channel coding parameters to the given channel conditions, thus providing better quality and lower latency of audio in home networking. [ABSTRACT FROM AUTHOR]

This paper introduces a promising property in which the transmitter of an OFDM system can utilize an N-point IFFT operation to build the time-domain symbols of a KN-point IFFT operation. Based on this property, we propose a flexible OFDM structure. This structure will support uplink applications which include many transmitters using different numbers of subcarriers and one receiver with just one IFFT/FFT module with the subcarriers of least common multiple (LCM) for the number of subcarriers of all the transmitters. Therefore it reduces the design complexities of the receiver and increases the compatible characteristics for consumer electronics (CE). Compared with the original OFDM system, the proposed system improves the performance of the packet error ratio (PER) and the bit error ratio (BER) performance by using the appropriate K. In addition, the proposed structure can be designed to enlarge the bandwidth efficiency of the system and improve the performance of channel estimation compared, with the OFDM/TDM system. [ABSTRACT FROM AUTHOR]

Multimedia services including IPTV require huge multimedia data transmission on the internet. In order to improve a throughput of data transmission, network coding has been proposed to use network bandwidth (=throughput) efficiently with error free channel assumption. However, when the channel is not error free, network coding may not increase the throughput. In this case, effective erasure recovery codes such as Raptor codes have to be used with the network coding. We present the advantage of Raptor codes combined with network coding for high throughput multimedia data transmission services like IPTV. For the applications allowing retransmission, Hybrid-ARQ combined with network coding is also considered. [ABSTRACT FROM AUTHOR]

In this paper, we present an FPGA-based fast image warping method by applying data parallelization schemes. The parallelization of accesses to pixels relieves not only latency problem of the warping, but also bandwidth requirements of off-chip memory. The LUT data parallelization scheme efficiently replaces parallel arithmetic operations with neither of increased memory size for LUT entries nor clock frequency. Two implementations with different characteristics prove the effectiveness and efficiency of the proposed method. [ABSTRACT FROM AUTHOR]

The staircase scheme requires minimum buffer storage among all the broadcasting schemes that provides the video services. It however does not always deliver the video data to users in time. This problem has been overcome in the conservative staircase scheme, but the scheme requires more resources especially the buffer storage. In the conservative staircase scheme, the video is divided into segments depending upon the number of video channels allocated to it by the video server. The bandwidth of a video channel is equal to the viewing rate of the video. The segments are further divided into subsegments and the video channels into subchannels in such a way that a subchannel transmits all the subsegments of a single segment. In this paper, we propose a new mechanism for utilizing the buffer storage efficiently in the conservative staircase scheme. In the proposed scheme, the segments transmitted by each of the video channels are arranged in decreasing order of their indices. By doing this, the data of the segments can be delayed without failing their availability to the users in time. By delaying downloading of the data, only that data is stored in the buffer storage at the user's site that is needed immediately and in very near future. This scheme not only provides the video services using less storage space but also requires less disk transfer rate as compared to the conservative staircase scheme. Furthermore, the complexity of this scheme is much less than that of the conservative staircase scheme because it does not require the segments to be divided into subsegments and the video channels into subchannels. [ABSTRACT FROM AUTHOR]

The channel bandwidth variation and the disconnection during handoff are the most critical problems which degrade the video quality in wireless video streaming. To cope with these problems, we propose an efficient video streaming method in this paper, which does not only dynamically adjust the video transmission rate based on the channel bandwidth, but also minimize the error propagation during handoff Firstly, the channel bandwidth of the mobile worldwide interoperability for microwave access, called WiMAX~ is estimated by analyzing channel parameters including the carrier to interference and noise ratio (CINR) and the handoff is detected by using the handoff occurrence message (HuM). Secondly, the streaming server adjusts the next transmission rate according to the estimated channel bandwidth to avoid the network congestion and performs the intra refresh method that inserts an intra frame (I-frame) right after handoff by using the HOM to reduce the error propagation effectively. Experimental results indicate that the proposed method can improve the performance of the video streaming over mobile WiMAX'. [ABSTRACT FROM AUTHOR]

This paper presents an ejJicient discrete-time bandpass sigma-delta (.ZA) modulator and digital in-phase /quadrature (I/Q) demodulator for multistandard wireless applications. The proposed bandpass IA modulator provides higher speed using advanced switched-capacitor resonators which are faster than the conventional ones. The test chip has been implemented in a 0.18 jim CMOS process and occupied with the active chip area of 0.16 mm2. The power consumption of the fabricated chip is 2.34 mWwith a 1.8 Vsupply voltage. The measured peak signal-to-noise ratios (SNR) are 34 dBfor 1.536 MHz (T-DMB), 26 dB for 5 MHz (UMTS), and 20 dB for 10MHz (WiBro) bandwidths, respectively. This paper also covers the simple and robust digital I/Q demodulator which has been realized using a field programmable gate array (FPGA) for digital signal processing'. [ABSTRACT FROM AUTHOR]

In this paper the VoD architecture for heterogeneous networks based on scalable wavelet video coding is proposed, with enhanced spatial scalability. The system consists of the clients and server (C/S). Motion estimation is performed on the resolution of the wavelet domain using hierarchical variable size block matching (DWT-HVSBM). All the vectors of different spatial resolutions are encoded into multiple-layer motion scalability. The sub-pixel accurate MCTF is applied in the temporal frames, and the three-dimensional wavelet coefficients are encoded in 3D EBCOT which is the extension of JPEG2000 code streaming. The server can adapt to the end users' requirements to provide the service over a heterogeneous network that has users with different bandwidths. The experiments show the performance of the codec is comparable to MC‗EZBC, and the video on demand network system can simultaneously support multiple heterogeneous users' demands. [ABSTRACT FROM AUTHOR]

In orthogonal frequency division multiplexing communication systems, the system performance may be degraded by the effect of inter-carrier interference. In this paper, a new scheme is proposed to cancel the effect of ICI. The system performances are evaluated based on carrier-to-interference ratio and bit error rate. The simulation results show that the effect of phase error with the proposed scheme is lighter than that with Zhao's scheme. Hence, the BER performance of proposed scheme is better than that of Zhao's scheme even in a large frequency offset. [ABSTRACT FROM AUTHOR]

The performance evaluation of communication protocols in broadband fixed wireless access (BFWA) networks needs the accurate description of the statistical properties of the packet error process. In BFWA networks mainly operating above 20 GHz, a clear line-of-sight (LOS) path is necessary between the base stations (BS) and the subscribers (S). Rain attenuation is the dominant fading mechanism at these frequencies. In this paper, a Markov-based method to estimate the packet loss rate (PLR) performance over a dynamically, due to propagation, varying LOS BFWA channel is presented and tested against simulation results. The proposed method is based on the first- order Markovian modeling of the rain attenuation process. The proposed method can serve as a performance evaluation tool of various modulation and coding schemes (MCS) and is also applicable even when adaptive fade mitigation techniques are employed in the BFWA network. [ABSTRACT FROM AUTHOR]

With the adaptive subcarrier allocation scheme, system performance could be guaranteed by leaving out the weak subcarriers. However, under a time-variant channel, the side information can not feedback to transmitter instantly and, then, system performance is degenerated. Error control coding techniques provide a way to improve the system performance. In this paper, a subcarrier allocation scheme combined different error control codings techniques including the repeat code and linear block codes is proposed to improve system performance. The efficiency of this technique is demonstrated by the simulation results. The proposed scheme has a better BER performance than that with the original subcarrier allocation scheme. [ABSTRACT FROM AUTHOR]

Ultra-wideband (UWB) systems use wide band-width of 3.1 to 10.6 GHz to provide ultra high data-rate for wireless applications. Implementation of UWB RF transceiver in the 3–5 GHz band (mandatory mode) makes big challenges that must be solved by circuit and system techniques. The proposed radio frequency(RF) transceiver, which is implemented in 0.13μm complementary metal-oxide semiconductor (CMOS) technology, is a part of a three-chip solution, but later it will be one chip solution which includes RF, modulation and demodulation (Modem), and medium access controller (MAC). The proposed WiMedia RF transceiver system can transmit and receive 4- high-definition (HD) channel streaming simultaneously. [ABSTRACT FROM AUTHOR]

The Interactive service platform(ISP) provides a whole solution for Interactive digital TV(DTV) from front-end to back-end. The ISP mainly consists of a data broadcast system (DBS) in front-end and an application execution system(AES) in back-ends. In Interactive DTV domain, Interactivity is obtained through Interactive applications that can be transmitted in broadcast networks by means of a specified carousel schema. Since additional DSM-CC messages are used to encapsulate applications, the carousel scheme introduces redundant information that leads to bandwidth redundancy in ISP. This paper presents a whole solution for interactive service platform in interactive DTV domain. The architectures of the DBS and AES and their functional elements are Introduced respectively. As the bath of Interactive application transmission, the enhanced data carousel(EDC) scheme derived from DVB data carousel is described in detail. Compared to DVB object carousel scheme, the SOC scheme introduces less redundant information, shows a better performance in bandwidth redundancy and presents to be more compact. Optimization strategies applying to broadcast application execution environment based on SOC scheme in ASS are also presented. The proposed platform has been Implemented and the experimental result shows a good performance. [ABSTRACT FROM AUTHOR]

In this paper, a minimum cost scheduling algorithm is proposed for high-quality video streaming in dynamic bandwidth allocation networks. It targets the problem of service cost minimization in optimal smoothing algorithms. Based on the bandwidth usage and the number of bandwidth renegotiations, service cost functions are defined for each transmission segment and a group of segments. The relative cost flags are also defined for each segment, which indicate the direction in which the segment is merged with less service cost. By inspecting the relative cost flags on the boundaries of segments, it is determined whether neighboring segments should be merged, until there is no reduction in the overall service cost. From the simulation results, the proposed algorithm is shown to merge neighboring segments in the direction minimizing the service cost. Compared with the optimal smoothing algorithm, the overall service cost is greatly reduced with a similar level of network resource usage and less transmission rate variability, especially for larger renegotiation cost parameters. [ABSTRACT FROM AUTHOR]

This paper presents a network adapted selective frame-dropping algorithm for streaming media over the Internet, which aims to address the problem of random packet losses resulted from network bandwidth mismatching. The basic idea is to first determine a sending window for each GOP according to its rendering time interval, and then selectively drop some frames with low priority to ensure that other more important frames in this GOP can be reliably delivered to the receiver within the time limitation of the GOP s sending window. For each GOP, the frame-dropping policies are determined in advance according to the transmitting results of its preceding GOP, and then slightly re-adjusted after each frame has been sent according to the current available network bandwidth. Experimental results show that the proposed algorithm can achieve free-error and fluent rendering effects even in the case of worse network conditions. [ABSTRACT FROM AUTHOR]

Consumer access networks are low bandwidth, especially in the up-stream direction. This asymmetry is heightened for real-time multimedia applications. The web wail is not acceptable when in a conversational interface. We describe the use of KPML, a protocol for transporting user stimulus. We illustrate the application of KPML to a typical consumer activity, accessing information sources over the telephone. We include results measuring KPML against other protocols for transporting user stimulus. [ABSTRACT FROM AUTHOR]

In this paper, we have present the hardware design of a high-speed descrambling engine for the Multi-Stream CableCARD device. The designed descrambling engine has been implemented based on a FPGA. The presented design's scrambling engine has a parallel processing structure that is extendable according to required input bandwidth. Especially we have verified that designed descrambling engine supported up to 200 Mbps input bandwidth. [ABSTRACT FROM AUTHOR]

As an effective technique for combating multipath fading and for high-bit-rate transmission over wireless channels, orthogonal frequency-division multiplexing (OFDM) is extensively used in modern digital television terrestrial broadcasting (DTTB) systems to support high performance bandwidth-efficient multimedia services. In this paper, a new robust timing recovery scheme is proposed for the time domain synchronous OFDM (TDS-OFDM) based DTTB systems. Simulations under different channel situations verify the efficiency of the proposed scheme. [ABSTRACT FROM AUTHOR]

Recently, there is an active research about a new emerging video coding standard, scalable video coding (SVC). SVC adopts a layered coding scheme to generate multi-layered bitstream for heterogeneous environments. In addition to the layered coding, SVC performs temporal prediction using the hierarchical B picture scheme in order to provide temporal scalable representation of input sequences. Since the coding order of the hierarchical B picture scheme is different from that of conventional temporal prediction schemes, the layer-switching algorithm considering the djfferent coding order has to be developed. In this paper, we first investigate the problems of layer switching in SVC and propose an algorithm that enables the SVC decoder to cope with layer switching caused by the bandwidth variation of networks. With the proposed algorithm, the SVC decoder can decode the time-varying sub-bitstream without modification of encoder. We confirm the proposed methods through the experiments. [ABSTRACT FROM AUTHOR]

3D content, such as X3D/VRML and MPEG-4, is transported over a lot of different networks to many different terminals. The differences in bandwidth and performance are typically coped with compressed multi-resolution content: e.g. MPEG-4 Wavelet Subdivision Surfaces (WSS). WSS geometry has a very good compression, but suffers from a high CPU overhead when it is used for fine-grained on-the-fly adaptations of the geometry in execution time regulated applications, e.g. networked interactive 3D games. This paper therefore presents a new content adaptation method for WSS geometry which drastically frees the CPU for other application tasks. The proposed method has a minimal impact on the execution time of the Graphical Processing Unit (10% increase) despite its extra memory requirements (68% compared to single resolution geometry) and its decreased post T&L cache efficiency. [ABSTRACT FROM AUTHOR]

Recent advances in storage and networking technology are rapidly changing users' TV viewing patterns. Personal Video Recorders (PVRs) enable people to watch what they want when they want by recording TV programs into local disks. In addition, PVRs can easily share TV contents over peer-to-peer networks once they are equipped with broadband connections. However, the current broadband bandwidth makes it difficult to support HD-quality TV content streaming on the Internet. In this paper, we propose a novel HD-quality streaming architecture for real-time broadcast content distribution. We develop two efficient scheduling schemes, matching a communication pair and switching /replication for system load-balancing, to further utilize network and disk bandwidth. We also evaluate buffer requirement to continue storing time-shifted programs without missing any broadcast data. By extensive simulation experiments, we show the effect of our proposed schemes and time-shifting policy. [ABSTRACT FROM AUTHOR]

In this paper, we investigate a receive diversity combining method for orthogonal frequency division multiplexing wireless local area network systems with cochannel interference. The scheme utilizes interference plus noise power (INP) to obtain the weighting factors for signal to interference plus noise power ratio (SINR) based maximum ratio combining (MRC). INP is estimated by using the correlation between the subchannel gains within the coherence bandwidth of the channel, which is caused by the limited delay spread of the multipath channel. Simulation results show that the SINR based MRC using the proposed INP estimator reveals greatly improved packet error rate performance compared to the signal-to-noise ratio based MRC that only considers white channel noise. [ABSTRACT FROM AUTHOR]

Video streaming over IP networks faces many challenges due to the varying network conditions such as time-varying packet loss and fluctuating bandwidth. In this paper, we focus on streaming scalable video over IP networks in order to enhance the received video quality. We present an adaptive video transmission scheme which combines unequal loss protection and layer selecting. In this scheme, we estimate the packet loss effects on video quality using model-based distortion estimation approach, both for the base layer and the enhancement layers. Based the estimated distortion, the scheme adaptively assigns the unequal amount of forward error correction to each layer and adjusts the sending rate by dynamically selecting the appropriate number of layers to transmit, according to the packet loss rate and the available network bandwidth. The scheme can provide a graceful degradation of received video quality as packet loss increasing and available bandwidth decreasing. Simulations results show that our scheme can provide reasonable video quality under varying network conditions. [ABSTRACT FROM AUTHOR]

The video oriented applications require considerably large amount of bandwidth. The bandwidth assigned to a video is very much dependent on its popularity. For popular videos, the bandwidth requirement is large. The popularity of a video is not static; it changes as the time passes. In that case, the bandwidth allocated to the video needs to be changed as it is a scarce resource, and at the same time, the user services should not get disrupted. In this paper, we propose seamless channel transition using the Bespoke scheme. During the transition, the old users do not get disrupted and the new users get same services as it was provided with the old channels. We have proposed an elegant technique for channel transition. Though this technique is used for the Bespoke scheme yet it can be adapted for any broadcasting scheme. [ABSTRACT FROM AUTHOR]

This paper presents an efficient joint disparity- motion estimation algorithm and a fast motion estimation algorithm in tin MPEG-4 compatible stereoscopic sequence codec for 3D broadcasting. In stereoscopic sequence coding, the critical problems to be solved are double bandwidth and computational complexity. However, the previous trials to reduce coding bits have increased computational complexity. In this paper, we try to implement fast algorithms for stereoscopic sequences while maintaining high coding efficiency. Joint disparity-motion estimation can increase coding efficiency and reduce the computational complexity of the stereoscopic codec by using the relationship between the disparity and motion fields. The disparity vectors are estimated by using the left and right motion vectors and the previous disparity vectors in each time frame. In order to obtain more accurate disparity vectors, we employ the spatial prediction process after joint estimation. From joint estimation and spatial prediction, we can obtain the accurately predicted disparity vectors and then increase the coding efficiency from those predicted vectors. Moreover, we propose a fast motion estimation algorithm which utilizes correlation between the motion vectors of the neighboring blocks. We confirmed that the PSNR of the proposed method is 1.0∼2.0dB higher than ones of conventional methods. In addition, the processing time is reduced by almost 1/10. [ABSTRACT FROM AUTHOR]

The harmonic scheme is reported to have the best performance for user's waiting time, however, it does not always deliver the video data in time to the viewers. To overcome this shortcoming, some new schemes have been proposed. They include cautious, quasi-harmonic, and polyharmonic schemes. The cautious and quasi-harmonic schemes deliver the video data in time by employing extra bandwidth, whereas in the polyharmonic scheme extra storage space is required in comparison to the original harmonic scheme. The polyharmonic scheme has better performance for user's waiting time in comparison to the cautious as well as the quasi-harmonic scheme. However, it requires a viewer to wait for integral multiple of time slots before starting viewing the video. Furthermore, it requires considerable amount of initial buffer storage. In this paper, the modified polyharmonic scheme has been proposed that provides the video data in time unlike the harmonic scheme. More importantly, it relaxes the condition making a viewer to wait for integral multiple of time slots before starting viewing. The viewer can get the services even after the video data is stored in the buffer storage for a fraction of the time slot The initial storage requirement is less than the polyharmonic scheme. [ABSTRACT FROM AUTHOR]

PN-sequence padding (PNP) TDS-OFDM transmission scheme has recently been proposed as an appealing alternative to the traditional Cyclic Prefix (CP) OFDM technology as it can provide significant improvement in the spectrum efficiency. In this paper, an iterative method is introduced to subtract the PN sequence. padding for the TDS-OFDM systems to mitigate the residual inter symbol interference (ISI) from the imperfect channel estimation, and the inherent channel estimation techniques are also performed to track the channel changes. Results show that this iterative algorithm can effectively remove the impact from the residual ISI on the slow-fading broadcast channels, even in the presence of very large channel delays in the single-frequency simulcast networks (SFN). [ABSTRACT FROM AUTHOR]

The insufficient bandwidth of SDRAM access has created a bottleneck in the performance of displaying and processing when used in previous design of the photo card reader. In this paper, we propose three ways to overcome this drawback First, we double the clock rate of the SDRAM operation to increase the amount of the memory bandwidth. Second, we use a dual port design of the SDRAM with a double buffer for the strip module to increase the usage efficiency of the bandwidth. Third, we also use a double buffer for the mem•ctrl module to increase the usage efficiency of the bandwidth. Using the extra gate counts of double FIFO buffers results in an increase of 3.3% from the previous system. Our new design has an improvement in the processing speed of about 4.4 times for displaying photos. [ABSTRACT FROM AUTHOR]

Much redundancy of the computation exists in the JPEG2 000 encoder, especially at the lower bit rate. This drawback results in the more encoding time and working memory size. In this paper, a fast and efficient rate control approach for JPEG2 000 is proposed, which can greatly reduce the computation operation and encoding time with nearly same performance as the conventional rate control scheme in JPEG2000 standard. [ABSTRACT FROM AUTHOR]

This paper compares the performance of coherent, differential and differential unitary space-time block codes operating over spatially correlated Rayleigh fading channels with up to four transmit antennas. It is shown that, when operating over spatially correlated channels, the relative performance of all candidate codes is improved when the number of transmit antennas is kept low. Improved performance is also observed when differential STBC are used compared to coherent STBC. [ABSTRACT FROM AUTHOR]

This paper presents an efficient method based on inter-field information for video de-interlacing. The basic idea is to classify the field dynamically to background or foreground area. The proposed method interpolates missing pixels using temporal information in the background area, and then interpolates remaining pixels using spatial and temporal information in the foreground area. Extensive simulations conducted for video sequences show that the performance of the proposed method is superior to the previous methods based on the ELA and the line averaging method in terms of the objective and subjective image qualities. Moreover, our proposed algorithm requires lower computational complexity and provides high-quality video sequences on the progressive devices. [ABSTRACT FROM AUTHOR]

The paper presents a new low complexity design of Psycho-A coustic Model (PAM), which is the key technology for MPEG-2/4 Advanced Audio Coding (AAC) encoding. The real-time constraint of MPEG AA C leads to a heavy computational bottleneck on today's portable devices. To overcome this problem, design analysis and optimization of PAM are addressed at the algorithmic level. First, the dominant calculation of spreading function is replaced with small look-up tables. Second, Modified-Discrete-Cosine-Transform-based (MDCT-based) PAM with block type decision in frequency domain is proposed to reduce the overall AAC encoder complexity and maintain the quality. This technique is different from the traditional methods. The computational complexity of PAM is reduced by 82% in total. The proposed design could be implemented in a real-time MPEG-2/4 AA C stereo encoder at Low Complexity profile and bitrate 128 kb/s with computational complexity of less than 20 MOPS while maintaining CD quality. [ABSTRACT FROM AUTHOR]

In this paper a video processing engine with accurate motion detection and sawtooth artifacts remove capabilities for LCD TV is proposed Since the accuracy of the motion detection is key factor to obtain excellent visual quality pictures, by incorporating the temporal and spatial information of video fields and sophisticated detection schemes, the goal can be obtained Also, a novel window-expanding search approach for low-angle edge detection/interpolation to remove the sawtooth artifacts, which occur on the edges/lines of fast moving images while doing the de-interlacing, is proposed To validate the proposed methods and consider the bandwidth requirement, an FPGA -based system for real-time emulation and verification is implemented And, from the real-time emulation results, it can be seen that the proposed algorithm has quite good performance in terms of obtaining good visual quality images. [ABSTRACT FROM AUTHOR]

Wireless multimedia streaming applications have been widely propagated with the rapid evolution of mobile handsets and wireless Internet technologies, especially MPEG-4 video streams over wireless communication networks. We propose an efficient methodology for Multimedia DRM which is applicable to the mobile terminal software. The methodology is comprised of the encryption methods on the media data that are encoded with MPEG-4 video format We constructed 3 methods, one is the way we can encrypt macro blocks (MBs) of I- VOP(Video Object Plane), another is P-VOP's MBs encryption and the other is the combination of these two methods. They are designed to DRM services for a wireless video streaming system without performance degradation that could be a burden to a mobile handset, where there is inherently preferred system that costs for low-power, low-network bandwidth. We report competitiveness of these methods on the perspective of performance evaluation with respect to the plain MPEG-4 files. [ABSTRACT FROM AUTHOR]

This paper provides a data placement method based on rate staggering to store scalable video data in a disk-array-based video server. Scalable video means a video which is coded in such a way that subsets of the full video bit stream can be decoded to create low quality/resolution videos. Supporting layered multiple resolutions from a video server is very desirable in many applications. Note that in a disk array the video data corresponding to different rates of the same video clip are not required to reside in the same disk In view of this, we propose and explore in this paper the approach of rate staggering, i.e., staggering video data in the disk array based on data rates. It is shown that the advantages of the proposed rate staggering method include: (1) minimizing the intermediate buffer space required by the server, (2) achieving better load balancing due to finer scheduling granularity, and (3) alleviating the disk bandwidth fragmentation. These advantages enable a video server using the rate staggering method to provide feasible solutions to some video stream requests which' cannot be met otherwise. The system throughput can thus be increased We also conduct several simulations for various applications. These experimental results show that the rate staggering method can significantly improve the performance of a VOD system. [ABSTRACT FROM AUTHOR]

The performance of a Pre-rake combiner (PRC) for a bi-polar Time-Hopping Ultra Wideband (TH UWB) system with simple receiver structure is studied in a realistic channel mode that is accepted by the IEEE 802.15.3a task group. A Pre-rake combiner con tributes to the boost of the desired signal and the mitigation of the inter-symbol interference (151). The significant improvement is accomplished from 10 to -2 BER by the pre-rake combining fingers. Master pre-rake combiner is installed in an access point counting the UWB channel characteristic. [ABSTRACT FROM AUTHOR]

Blind equalization of communication channel is important because it does not need training signal, nor does it require a priori channel information. So, we can increase the bandwidth efficiency. The linear prediction error method is perhaps the most attractive in practice due to the insensitive to blind channel equalizer length mismatch as well as for Us simple adaptive implementation. Unfortunately, the previous one-step prediction error method is known to be limited in arbitrary decision delay. In this paper, we propose method for fractionally spaced blind equalizer with arbitrary decision delay using one-step forward prediction error filter from second-order statistics of the received signals for SIMO channel. Our algorithm utilizes the forward prediction error as training signal and computes the best decision delay from all possible decision delay. Simulation results are presented to demonstrate the performance of our proposed algorithm. [ABSTRACT FROM AUTHOR]

This paper concerns the power splitter for CATV transmission system including down-converted DBS signals. In this work, a novel transformer-type Wilkinson power splitter with a compensating circuit was developed for application in a CATV transmission system. In order to extract the best RF performance form the power a splitter, it was optimally designed using a newly proposed design method that employs transformer-type lumped equivalent circuit The power splitter showed highly improved RF characteristics in comparison with conventional ones. It exhibited a bandwidth that extended up to 1500 MHz, and return loss and isolation values lower than -15 dB from 5 MHz to 1500 MHz. The above results indicate that the transformer-type Wilkinson power splitter with a compensating circuit is a promising candidate for application in CATV [ABSTRACT FROM AUTHOR]

This paper addresses the implementation architecture and protocol procedure of the dedicated broadband access network TV transmission system. According to application-oriented quality of service (QoS), two alternative rate control schemes for multiple DTV programs transmission are proposed collaborating with MPEG-2 system strategy and IP-base network multiplexing mechanism. As to the network bandwidth utilization, a flexible RTP-based synchronization resilience scheme for VBR transport stream (TS) transmission is realized to ensure the performance requirement for MPEG-2 standard As to the MPEG-2 system strategy, moreover, a parallel VBR transcoding/encoding rate control algorithm for CBR TS transmission could achieve a consistent picture quality for multiple DTV programs under a constrained fixed-rate network bandwidth. For adapting the variable access network bandwidth, the system main control module can flexibly schedule the available rate control scheme for multiple slave service multiplexers. Experimental results demonstrate that the proposed schemes not only improve the traditional TV broadcast adaptation to Internet, but also achieve a more ideal program picture quality constrained by multi-program parallel multiplexing. [ABSTRACT FROM AUTHOR]

Frequency modulation (FM) is widely used in applications such as FM broadcasting, Global System for Mobile (GSM whose modulation method, Gaussian Minimum Shift Keying, is a kind of FM) etc. Using digital signal processing (DSP) to demodulate FM signals has advantages over analogue methods in respect of the easiness of design, adjustment, stability, reliability, and upgrading. The current digital methods employ an even spaced (uniform) sampling method and process each sample. When the modulation index increases, the modulated signal bandwidth expands. So the sampling rate needs to be increased as well. This leads to increased computation time especially for division and arctangent. In this paper, we propose a non-uniform low-sampling method, whose overall sampling rate is not a function of the modulation index. Its minimum rate can be as small as four times the bandwidth (B) of the non-modulated (message) signal. Furthermore, the minimum rate of division and arctangent computations are only 2B Hz. As a result, the proposed method can significantly reduce the overall computation for FM demodulation. [ABSTRACT FROM AUTHOR]

As an effective technique for combating multipath fading and for high data rate transmission over wireless channels, orthogonal frequency division multiplexing (OFDM) is extensively used in wireless local area network (WLAN) systems to support high performance bandwidth-efficient multimedia services. In this paper, novel synchronization and channel estimation schemes are proposed for the OFDM- based WLAN systems. Analyses and simulation results show that the proposed schemes exhibit good performances and are robust in harsh fading channel environments and low signal- to-noise ratio (SNR) conditions. [ABSTRACT FROM AUTHOR]

Transcoding is a core technique that is used in providing quality-of-service (QoS) adaptive video-on-demand service. Many studies have examined how best to perform transcoding and reduce computation overhead. However, the question of when to transcode has not been adequately studied in previous research. This paper addresses this question and presents a simple and intelligent approach that can be used to reduce both disk bandwidth and space requirements. Our approach determines the optimum time to apply transcoding by considering the potential benefits that can be realized. For instance, in order to save disk bandwidth for frequently accessed content, it pre-creates and stores multiple QoS versions. On the other hand, in order to save disk space for rarely accessed content, it stores only a single QoS version and performs transcoding on the fly. The key is to find the optimal threshold between pre-created multiple QoS versions and on-demand transcoding. We compute the optimal threshold by using a mathematical model. A simulation-based experiment to evaluate the effectiveness of our new approach highlights three advantages. First, our method effectively reduces both disk bandwidth and space requirements. Second, our technique is more efficient for skewed access patterns. Third, the threshold computed by our mathematical model results in improved performance regardless of environmental parameters. [ABSTRACT FROM AUTHOR]

MPEG-AA C is the current state of the art in audio compression technology. The CD-quality promised at bit rate as low as 64 kbps makes AAC a strong candidate for high quality low bandwidth audio streaming applications over wireless network. Besides this low bit rate requirement, the codec must be able to run on personal wireless handheld devices with its inherent low power characteristics. While the AA C standard is definite enough to ensure that a valid AA C stream is correctly decodable by all AA C decoders, it is flexible enough to accommodate variations in implementation, suited to different resources available and application areas. This paper reviews various implementation techniques of the encoder. We then proposed our method of an optimized software implementation of MPEG-AAC (LC profile). The coder is able to perform encoding task using half the processing power compared to standard implementation without significant degradation in quality as shown by both subjective listening test and an ITU-R compliant quality testing program (OPERA). [ABSTRACT FROM AUTHOR]

The client's buffer requirement, the server network bandwidth, and the access latency are three key parameters for the design of Video-on-Demand (Vod) system. Many schemes have been proposed to provide VoD service, one of which is the Mirrored-Pyramid Broadcasting (MPB) scheme. However, it is not optimal with regard to the client's buffer requirement, the bandwidth utilization, and the access latency. After discussing the drawbacks of the MPB scheme, we propose an Advanced Mirrored-Pyramid Broadcasting (AMPB) scheme, which has a novel channel design and a novel broadcast schedule. Experimental results show that AMPB saves up to 75% of the client's buffer and 50% of the server network bandwidth. It also achieves a better access latency. [ABSTRACT FROM AUTHOR]

Narrowband interference (NBI) is one of the most important problems in an OFDM system, which not only degrades the signal performance on the overlapped subcarriers, but also affects the signal performance on the nearby subchannels due to the spectral leakage effect of DFT demodulation at the receiver. In this paper, we consider the NBI problem caused by narrow band digital communication systems, and propose a novel NBI cancellation method We consider the NBJ reconstruction problem including NBI central frequency and its bandwidth estimation. Simulation results show that this method can achieve about 6dB signal to interference plus noise ratio (SINR) gain if the interference bandwidth is about twice of that OFDM bins and the system SINR is about 0 dB. As the bandwidth of NBI decreases, more SINR gain will be obtained. Meanwhile, we also consider the system implementation problem, it has shown that its implementation complexity is not high based on the current digital signal processing techniques. If zoom FFT is employed in our system, the calculation complexity can be reduced considerably. [ABSTRACT FROM AUTHOR]

Channel variation during an OFDM block leads to the loss of orthogonality among subcarriers, resulting in inter-carrier interference (ICI) in orthogonal frequency division multiplexing (OFDM) systems. Many schemes have been proposed to suppress ICL but they are computationally complex or at the price of sacrificing bandwidth. In some cases, such as high-density television (HDTV) broadcasting and satellite OFDM systems, the very long delay spreads pose the possibility that the channel length exceeds that of the moderate cyclic prefix (CP), resulting in intersymbol interference (ISI) and intercarrier interference (WI). A time- domain equalizer (TEQ) is usually used in the receiver to reduce the duration of the overall response of the transmission system, and therefore minimize the ISI and ICI However, the optimum design of TEQ turns out to be a difficult task The ICI's introduced by insufficient CP and channels variations are different and usually exploit different methods to mitigate them. In this paper, we provide an ICI and ISI analysis in time- and frequency- domain for OFDM systems with insufficient CP over time-variant channels. Basing on this analysis, we propose an iterative method for joint ICI and ISI cancellation whose complexity is linear in the OFDM symbol length. Theoretical analysis and simulation results indicate that the proposed method can effectively mitigate ICI and ISI introduced by channel variation and insufficient CP with high bandwidth efficiency. [ABSTRACT FROM AUTHOR]