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Broadcasting, IEEE Transactions on

Issue 3 • Date Sep 2002

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Displaying Results 1 - 9 of 9
  • Time domain phase noise correction for OFDM signals

    Publication Year: 2002 , Page(s): 230 - 236
    Cited by:  Papers (35)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (276 KB) |  | HTML iconHTML  

    We introduce an algorithm for compensating for carrier phase noise in an OFDM communication system. Through the creation of a linearized parametric model for phase noise, we generate a least squares (LS) estimate of the transmitted symbol. Using digitized DVB-T RF signals created in a laboratory and a DVB-T compliant receiver model, simulation results are presented to evaluate the effectiveness of the algorithm in practical environments. View full abstract»

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  • An outband paging protocol for energy-efficient mobile communications

    Publication Year: 2002 , Page(s): 246 - 256
    Cited by:  Papers (6)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (621 KB) |  | HTML iconHTML  

    A new protocol is proposed for reducing the power consumption of battery-powered terminals in a mobile computing environment. We exploit the fact that, in a mobile data network, mobile terminals do not continuously receive data and therefore they need not continuously operate their receivers. Nevertheless, they need to check their traffic condition periodically, that is, whether there are pending data for them or not. The proposed energy-efficient protocol is based on a paging procedure wherein a dedicated channel is used to alert (page) terminals with pending traffic. Each terminal may check its traffic condition whenever it decides to by monitoring the paging channel. The protocol is evaluated through an approximated theoretical model and through computer simulation. We focus on deriving approximate formulas for the mean message delay, the message delay variance and the power consumption. It is shown that the proposed protocol can achieve considerable power saving at a cost of increased message delivery delay. View full abstract»

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  • Achieving inter-session fairness for layered video multicast

    Publication Year: 2002 , Page(s): 215 - 222
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (277 KB) |  | HTML iconHTML  

    The Internet is increasingly used to deliver multimedia services. Since there are heterogeneous receivers and changing network conditions, it has been proposed to use adaptive rate control techniques such as layered video multicast to adjust the video traffic according to the available Internet resources. A problem of layered video multicast is that it is unable to provide fair bandwidth sharing between competing video sessions. We propose two schemes, layered video multicast with congestion sensitivity and adaptive join-timer (LVMCA) and layered video multicast with priority dropping (LVMPD), to achieve inter-session fairness for layered video multicast. Receiver-driven layered multicast (RLM), layer-based congestion sensitivity, LVMCA, and LVMPD are simulated and compared. Results show both proposed schemes, especially LVMPD, are fairer and have shorter convergence time than the other two schemes. View full abstract»

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  • Variable shortened-and-punctured Reed-Solomon codes for packet loss protection

    Publication Year: 2002 , Page(s): 237 - 245
    Cited by:  Papers (11)  |  Patents (6)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (330 KB) |  | HTML iconHTML  

    Techniques using Reed-Solomon (RS) codes to recover lost packets in digital video/audio broadcasting and packet switched network communications are reviewed. Usually, different RS codes and their corresponding encoders/decoders are designed and utilized to meet different requirements for different systems and applications. We incorporate these techniques into a variable RS code and present encoding and decoding algorithms suitable for the variable RS code. A mother RS code can be used to produce a variety of RS codes and the same encoder/decoder can be used for all the derivative codes, with adding/detecting zeros, removing some parity symbols and adding erasures. A VLSI implementation for erasure decoding of the variable RS code is described and the achievable performance is quantitatively analyzed. A typical example shows that the signal processing speed is up to 2.5 Gbits/second and the processing delay is less than one millisecond, when integrating the decoder on a single chip. Therefore, the proposed algorithm and the encoder/decoder can universally be utilized for different applications with various requirements, such as transmission data rate, packet length, packet loss protection capacity, as well as layered protection and adaptive redundancy protection in DVB/DAB, Internet and mobile Internet communications. View full abstract»

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  • Multistream transmission for hybrid IBOC-AM with embedded/multidescriptive audio coding

    Publication Year: 2002 , Page(s): 179 - 192
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (380 KB) |  | HTML iconHTML  

    Hybrid in band on channel (IBOC) digital audio broadcasting simultaneously with analog amplitude modulation (AM) has been proposed as a solution to DAB in the AM band. Since the AM band is crowded and the available bandwidth per program is limited, adding digital transmission presents a challenge. To achieve FM-like audio quality, an audio coder rate of 32-64 kb/sec may be required. One currently proposed hybrid IBOC-AM system is 30 kHz wide. Severe second adjacent interference may occur in certain areas, leading to a possible 40% loss of the effective audio bit rate. To cope with such harsh transmission conditions, we present a solution based on embedded/multidescriptive audio coding with matched multistream transmission in separate frequency bands. With the loss of one frequency band, the embedded system blends to a lower audio coder rate with a much better quality than analog AM. The nonembedded system, without multistream transmission, fails catastrophically when a little more than one sideband is severely interfered with, causing a severe discontinuity in quality while blending directly to analog AM. Some detailed robust embedded systems are outlined. We also show how multistream transmission schemes can be used with nonembedded audio coders. Both daytime and nighttime scenarios are included. This paper contains a catalog of possible systems for different audio quality levels and interference scenarios, including systems with 20 kHz bandwidth rather than 30 kHz. View full abstract»

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  • Effective multi-program broadcasting of prerecorded video using VBR MPEG-2 coding

    Publication Year: 2002 , Page(s): 207 - 214
    Cited by:  Papers (9)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (296 KB) |  | HTML iconHTML  

    The tradeoff between picture quality and bandwidth usage is a prominent issue in the world of broadcasting. Since broadcasters are able to transmit multiple streams simultaneously in a channel, they face the challenge of guaranteeing the contracted picture quality required by each of the transmitted video streams while maximizing the number of video streams carried in each channel. We have developed an easy to implement MPEG-2 based multi-program video coding system suitable for digital TV broadcast, video on demand, and high definition TV over broadcast satellite networks with limited bandwidth. Compared to present broadcast systems and for the same level of contracted picture quality, our system greatly increases the number of video streams transmitted in each channel. As a result, either a large number of transponders can be freed to carry real-time broadcasting or the level of picture quality can be significantly increased. By switching from tape storage to video server technology, the need for numerous playback (VTR) systems at the headend is eliminated. In addition, the most of the complete MPEG-2 encoders are replaced by much less complex MPEG-2 transcoders. All this means a much more cost-effective solution for broadcast stations. View full abstract»

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  • Huffman code based error screening and channel code optimization for error concealment in perceptual audio coding (PAC) algorithms

    Publication Year: 2002 , Page(s): 193 - 206
    Cited by:  Papers (4)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (456 KB) |  | HTML iconHTML  

    The class of perceptual audio coding (PAC) algorithms yields efficient and high-quality stereo digital audio bitstreams at bit rates from 16 kb/sec to 128 kb/sec (and higher). To avoid "pops and clicks" in the decoded audio signals, channel error detection combined with source error concealment, or source error mitigation, techniques are preferred to pure channel error correction. One method of channel error detection is to use a high-rate block code, for example, a cyclic redundancy check (CRC) code. Several joint source-channel coding issues arise in this framework because PAC contains a fixed-to-variable source coding component in the form of Huffman codes, so that the output audio packets are of varying length. We explore two such issues. First, we develop methods for screening for undetected channel errors in the audio decoder by looking for inconsistencies between the number of bits decoded by the Huffman decoder and the number of bits in the packet as specified by control information in the bitstream. We evaluate this scheme by means of simulations of Bernoulli sources and real audio data encoded by PAC. Considerable reduction in undetected errors is obtained. Second, we consider several configurations for the channel error detection codes, in particular CRC codes. The preferred set of formats employs variable-block length, variable-rate outer codes matched to the individual audio packets, with one or more codewords used per audio packet. To maintain a constant bit rate into the channel, PAC and CRC encoding must be performed jointly, e.g., by incorporating the CRC into the bit allocation loop in the audio coder. View full abstract»

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  • Channel estimation techniques based on pilot arrangement in OFDM systems

    Publication Year: 2002 , Page(s): 223 - 229
    Cited by:  Papers (482)  |  Patents (67)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (282 KB) |  | HTML iconHTML  

    Channel estimation techniques for OFDM systems based on a pilot arrangement are investigated. Channel estimation based on a comb type pilot arrangement is studied through different algorithms for both estimating the channel at pilot frequencies and interpolating the channel. Channel estimation at pilot frequencies is based on LS and LMS methods while channel interpolation is done using linear interpolation, second order interpolation, low-pass interpolation, spline cubic interpolation, and time domain interpolation. Time-domain interpolation is obtained by passing to the time domain by means of IDFT (inverse discrete Fourier transform), zero padding and going back to the frequency domain by DFT (discrete Fourier transform). In addition, channel estimation based on a block type pilot arrangement is performed by sending pilots in every sub-channel and using this estimation for a specific number of following symbols. We have also implemented a decision feedback equalizer for all sub-channels followed by periodic block-type pilots. We have compared the performances of all schemes by measuring bit error rates with 16QAM, QPSK, DQPSK and BPSK as modulation schemes, and multipath Rayleigh fading and AR based fading channels as channel models. View full abstract»

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  • A DAB transmitter prototype with high flexibility and low cost

    Publication Year: 2002 , Page(s): 173 - 178
    Cited by:  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (236 KB)  

    The design and implementation of a DAB transmitter prototype is proposed. The system architecture, ensemble multiplexer, channel encoder, OFDM (orthogonal frequency division multiplexing) modulator and other digital baseband parts are described. The prototype is fully compatible with ETS 300 401 and other related standards. It supports different services with various data formats and provides a convenient PC-based user interface. All I/Os and core parts are based on a modular design with a universal internal interface, hence the prototype can be flexibly configured to adapt to different requirements. Compared with existing products, the proposed transmitter prototype offers higher flexibility and lower cost. View full abstract»

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Aims & Scope

IEEE Transactions on Broadcasting covers the field of broadcast technology, including the production, distribution, transmission, and propagation aspects of broadcasting.

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Editor-in-Chief
Yiyan Wu
Communications Research Ctr Canada