Compatibility of FM hybrid in-band on-channel (IBOC) system fordigital audio broadcasting

A robust in-band on-channel (IBOC) digital audio broadcast (DAB) system for improved performance over existing FM broadcasting is under development by Westinghouse for USA Digital Radio. Both the analog FM and the DAB signals are transmitted simultaneously in the FM hybrid IBOC system. Broadcasters can simultaneously transmit both analog and digital signals within the allocated channel mask, allowing full compatibility with existing analog receivers. It is shown here that the solution is tolerant of interference from adjacent channels, or interference from the co-channel analog transmission, even in a multiple station, strong-signal urban market. Although the primary focus of this paper is to discuss the compatibility issues between the existing FM and the DAB signals, the paper also briefly describes spectral occupancy, power ratios, modulation formats, and coding, as well as the introduction of frequency and time diversity     

Homogeneous distribution of frequencies in a regular hexagonal cell system

A theory of optimal adjacent channel distance in homogeneous hexagonal cell systems is developed. Algorithms generating frequency distributions with best adjacent channel distance, valid for arbitrary co-channel reuse ratios, are presented. The theory may be applied to mobile radio telephone systems or broadcast services.     

An integrated error correction and detection system for digitalaudio broadcasting

Hybrid in-band on-channel digital audio broadcasting systems deliver digital audio signals in such a way that is backward compatible with existing analog FM transmission. We present a channel error correction and detection system that is well-suited for use with audio source coders, such as the so-called perceptual audio coder (PAC), that have error concealment/mitigation capabilities. Such error mitigation is quite beneficial for high quality audio signals. The proposed system involves an outer cyclic redundancy check (CRC) code that is concatenated with an inner convolutional code. The outer CRC code is used for error detection, providing flags to trigger the error mitigation routines of the audio decoder. The inner convolutional code consists of so-called complementary punctured-pair convolutional codes, which are specifically tailored to combat the unique adjacent channel interference characteristics of the FM band. We introduce a novel decoding method based on the so-called list Viterbi algorithm (LVA). This LVA-based decoding method, which may be viewed as a type of joint or integrated error correction and detection, exploits the concatenated structure of the channel code to provide enhanced decoding performance relative to decoding methods based on the conventional Viterbi algorithm (VA). We also present results of informal listening tests and other simulations on the Gaussian channel. These results include the preferred length of the outer CRC code for 96-kb/s audio coding and demonstrate that LVA-based decoding can significantly reduce the error flag rate relative to conventional VA-based decoding, resulting in dramatically improved decoded audio quality. Finally, we propose a number of methods for screening undetected errors in the audio domain     

64 kbit/s audio signal transmission approaches using 32 kbit/sADPCM channel banks

Two simple 64-kb/s wideband coding approaches using 32-kb/s ADPCM (adaptive digital pulse-code modulated) channel banks are proposed and compared to CCITT 64 kb/s ADPCM, which is being recommended as CCITT G.722. These two, folding ADPCM and QMF ADPCM, are intended to pave the way for smooth transition from conventional 4-kHz band telephone systems to 7-kHz wideband systems in private networks. The first approach, supporting the high-quality audio program transmission, requires only samplers and multiplexers at the input and output ports of the channel banks. In the second approach, samplers and multiplexers are replaced by quadrature mirror filters in order to increase coding quality. Performance test results for audio signal transmission show that these simplified approaches provide an inexpensive way to introduce wideband communication systems     

Multilevel RS/convolutional concatenated coded QAM for hybridIBOC-AM broadcasting

Bandwidth efficient modulation schemes using Reed-Solomon (RS) codes are proposed for hybrid in-band-on-channel (IBOC) systems that broadcast digital audio signals simultaneously with analog amplitude modulation (AM) programs in the AM band. Since both the power and bandwidth allocated for digital audio transmission are limited in this application, the system cannot afford to add enough redundancy for error control using conventional concatenated coding schemes. We show that by using multilevel RS and convolutional concatenated coded quadrature amplitude modulation (QAM), an efficient modulation schemes can be obtained for applications such as IBOC-AM broadcasting     

An OFDM all digital in-band-on-channel (IBOC) AM and FM radiosolution using the PAC encoder

The advances in digital communications and compression algorithms have made more efficient and more robust transmission schemes possible. Radio broadcast systems have not fully utilized these advances to their benefit. All digital robust radio broadcast systems for the AM and the FM band are proposed. The proposed systems are based on orthogonal frequency division multiplexing (OFDM) technology in conjunction with PAC for both the AM and the FM bands. The Perceptual Audio Coder (PAC) developed by Bell Laboratories compresses audio signals very efficiently with CD-like quality at 96 kbps and stereo FM-like quality at 48 kbps. These are rates achievable with 200 kHz and 30 kHz bandwidths available per FM and AM station respectively. No new spectrum is required since the digital signals are transmitted within the current allocated FCC masks. In an FM channel, a wide-band data subchannel can be provided in addition to the 96 kbps error protected PAC audio information     

A novel ARQ technique using the turbo coding principle

A novel automatic repeat request (ARQ) technique based on the turbo coding principle is presented. The technique uses the log-likelihood ratios generated by the decoder during a previous transmission as a priori information when decoding retransmissions. Simulation results show a significant decrease in frame error rate, especially at low-to-moderate E_b/N₀     

Joint encoding and decoding methods for digital audio broadcasting of multiple programs

We introduce new methods for increasing the performance of multiprogram digital audio broadcast systems, e.g., satellite digital audio broadcasting. Joint multiprogram encoding is an attractive possibility for parallel broadcasting of a large number of programs. Joint coding extended over multiple audio frames in time give further improvements. The benefits of this kind of statistical multiplexing yield improved audio quality and/or higher capacity in terms of number of programs. We describe the new Joint Multiple Program Encoding Technique in the context of the perceptual audio coding (PAC) type of algorithms. We also describe methods for multi-program transmission including Equal Error Protection (EEP) as well as Unequal Error Protection (UEP) and improved error concealment for multiple program transmission. Some of the techniques described in this paper, are currently being used in satellite digital audio broadcasting in the United States.     

Co-Channel and Adjacent Channel Interference in Wireless Cellular Communications

In this paper, the impact of co-channel and adjacent channel interference in mobile cellular radio systems on the cellular system capacity are studied. Simple empirical expressions for average signal to co-channel, signal to adjacent channel and signal to total interference ratios are obtained. The simple empirical results developed in this paper are easier to use in a cellular system design. The maximum number of bits/sec that can be transmitted over a cell area for a given bit error rate and total bandwidth, is proposed as a new definition of capacity of digital cellular systems. The capacity of a cellular system is shown to depend on the excess bandwidth and the number of active users.     

The EIA digital audio sub-committee VHF channel characterizationtest

The Electronic Industries Association (EIA) DAR subcommittee is evaluating proposed digital audio radio (DAR) systems for the United States. As a part of that process, the subcommittee is conducting a “VHF radio channel characterization test”. The results of the test will be applied directly to the laboratory testing of the DAR systems and will also find general use within the radio communications industry     

调频段DRM数字广播信道技术研究

2009年8月ETSI颁布的DRM(Digital Radio Mondiale)标准将调频段DRM数字广播囊括其中,介绍该标准中调频段DRM数字广播的优势与信道部分的技术特性。调频段DRM数字广播通过信道编码技术、信道复用、OFDM调制等技术,实现调频段广播信号的传输,与DAB相比可快速而平滑地实现FM波段广播由模拟向数字的过渡。     

CD3-OFDM: a novel demodulation scheme for fixed and mobilereceivers

This paper describes a novel channel estimation scheme identified as coded decision directed demodulation (CD3) for coherent demodulation of orthogonal frequency division multiplex (OFDM) signals making use of any constellation format [e.g., quaternary phase shift keying (QPSK), 16-quadrature amplitude modulation (QAM), 64-QAM]. The structure of the CD3-OFDM demodulator is described, based on a new channel estimation loop exploiting the error correction capability of a forward error correction (FEC) decoder and frequency and time domain filtering to mitigate the effects of noise and residual errors. In contrast to the conventional coherent OFDM demodulation schemes, CD3-OFDM does not require the transmission of a comb of pilot tones for channel estimation and equalization, therefore yielding a significant improvement in spectrum efficiency (typically between 5-15%). The performance of the system with QPSK modulation is analyzed by computer simulations, on additive white Gaussian noise (AWGN) and frequency selective channels, under static and mobile reception conditions. For convolutional coding rate 1/2, the results indicate that CD3-OFDM allows one to achieve a very fast adaptation to the channel characteristics in a mobile environment (maximum tolerable Doppler shift of about 80 Hz for an OFDM symbol duration of 1 ms, as differential demodulation) and an E_b /N₀ performance similar to coherent demodulation (e.g., E_b/N₀=4.3 dB at bit-error rate (BER)=2·10 ^-4 on the AWGN channel). Therefore, CD3-OFDM can be suitable for digital sound and television broadcasting services over selective radio channels, addressed to fixed and vehicular receivers     

List Viterbi algorithms for continuous transmission

The conventional list Viterbi algorithm (LVA) produces a list of the L best output sequences over a certain block length in decoding a terminated convolutional code. We show in this paper that the LVA with a sufficiently long list is an optimum maximum-likelihood decoder for the concatenated pair of a convolutional code and a cyclic redundancy check (CRC) block code with error detection. The CRC is used to select the output. New LVAs for continuous transmission are proposed and evaluated, where no termination bits are required for the convolutional code for every CRC block. We also present optimum and suboptimum LVAs for tailbiting convolutional codes. Convolutional codes with Viterbi decoding were proposed for so-called hybrid in band on channel (hybrid IBOC) systems for digital audio broadcasting compatible with the frequency modulation band. For high-quality audio signals, it is beneficial to use error concealment/error mitigation techniques to avoid the worst type of channel errors. This requires a reliable error flag mechanism (error detection feature) in the channel decoder. A CRC on a block of audio information bits provides this mechanism. We demonstrate how the LVA can significantly reduce the flag rate compared to the regular Viterbi algorithm (VA) for the same transmission parameters. At the expense of complexity, a receiver optional LVA can reduce the flag rate by more than an order of magnitude. The difference in audio quality is dramatic. The LVA is backward compatible with a VA     

An adaptive coding scheme with code combining for mobile radiosystems

The authors propose and study an adaptive error-control coding scheme for binary digital FM (BFM) mobile radio transmission. The scheme employs code combining through packet retransmissions. The number of transmissions of a packet is in proportion to the channel fading/noise levels, which is in contrast to time diversity techniques where a fixed number of repetitions of a data packet is performed even in the absence of channel errors. Furthermore, the receiver uses received signal envelopes as channel state information, which significantly improves the throughput and bit error rate (BER) performance. Performance of the proposed scheme is analyzed for frequency-flat Rayleigh fading channels with additive white Gaussian noise (AWGN), co-channel interference and random FM noise     

The Cost of Uncorrelation and Noncooperation in MIMO Channels

We investigate the capacity loss for using uncorrelated Gaussian input over a multiple-input multiple-output (MIMO) linear additive-noise channel. We upper-bound the capacity loss by a universal constant C* which is independent of the channel matrix and the noise distribution. For a single-user MIMO channel with n_t inputs and n_r outputs C* = min [ 1/2, n_r/n_t log₂ (1+n_t/n_r) ] bit per input dimension (or 2C* bit per transmit antenna per second per hertz), under both total and per-input power constraints. If we restrict attention to (colored) Gaussian noise, then the capacity loss is upper-bounded by a smaller constant C_G = n_r/2n_r log₂ (n_t/n_r) for n_r ges n_t/e, and C_G = 0.265 otherwise, and this bound is tight for certain cases of channel matrix and noise covariance. We also derive similar bounds for the sum-capacity loss in multiuser MIMO channels. This includes in particular uncorrelated Gaussian transmission in a MIMO multiple-access channel (MAC), and "flat" Gaussian dirty-paper coding (DPC) in a MIMO broadcast channel. In the context of wireless communication, our results imply that the benefit of beamforming and spatial water-filling over simple isotropic transmission is limited. Moreover, the excess capacity of a point-to-point MIMO channel over the same MIMO channel in a multiuser configuration is bounded by a universal constant.     

A Turbo Detection and Sphere-Packing-Modulation-Aided Space-Time Coding Scheme

A recently proposed space-time block-coding (STBC) signal-construction method that combines orthogonal design with sphere packing (SP), referred to here as STBC-SP, has shown useful performance improvements over Alamouti's conventional orthogonal design. In this contribution, we demonstrate that the performance of STBC-SP systems can be further improved by concatenating SP-aided modulation with channel coding and performing demapping as well as channel decoding iteratively. We also investigate the convergence behavior of this concatenated scheme with the aid of extrinsic-information-transfer charts. The proposed turbo-detected STBC-SP scheme exhibits a "turbo-cliff" at E_b/N₀=2.5 dB and provides E_b/N₀ gains of approximately 20.2 and 2.0 dB at a bit error rate of 10 ^-5 over an equivalent-throughput uncoded STBC-SP scheme and a turbo-detected quadrature phase shift keying (QPSK) modulated STBC scheme, respectively, when communicating over a correlated Rayleigh fading channel.A condensed version of this paper was presented at VTC'04 Fall, LA, USA     

Broadcasting with digital audio

Digital audio broadcasting will come on-line worldwide in the next few years. The technology-also known as digital audio radio or digital sound broadcasting-promises to provide sound of compact-disk quality, nearly free from multipath distortion or other transmission interferences. And digital audio broadcasting (DAB) is not just for sound: all sorts of information, digitally encoded, will be transmitted. Testing is already under way around the world for systems that deliver DAB signals from satellites, from terrestrial systems using newly assigned spectral bands, and from in-band (that is, currently assigned) AM and FM systems. The author discusses spectra for DAB, perceptual coding, Europe's Eureka 147 system, and DAB research in Canada, the USA, and Japan     

Technical advances in digital audio radio broadcasting

The move to digital is a natural progression taking place in all aspects of broadcast media applications from document processing in newspapers to video processing in television distribution. This is no less true for audio broadcasting which has taken a unique development path in the United States. This path has been heavily influenced by a combination of regulatory and migratory requirements specific to the U.S. market. In addition, competition between proposed terrestrial and satellite systems combined with increasing consumer expectations have set ambitious, and often changing, requirements for the systems. The result has been a unique set of evolving requirements on source coding, channel coding, and modulation technologies to make these systems a reality. This paper outlines the technical development of the terrestrial wireless and satellite audio broadcasting systems in the U.S., providing details on specific source and channel coding designs and adding perspective on why specific designs were selected in the final systems. These systems are also compared to other systems such as Eureka-147, DRM, and Worldspace, developed under different requirements.