Simultaneous broadcasting of analog FM and digital audio signals bymeans of adaptive precanceling techniques

A method for broadcasting digital audio simultaneously with existing analog frequency modulation (FM) radio is presented. The method is based on precomputing the response of the host analog FM signal at the digital receiver and precanceling it at the transmitter. As a result, the digital transmission is free from interference from analog FM. We select the rate and power level of the digital transmission in a manner that the interference the digital data incur on the analog FM signal remains at acceptably low levels. The digital transmission is based on adaptive orthogonal frequency-division multiplexing (OFDM) (adaptive multicarrier). The frequencies and number of carriers of the digital multicarrier modem are judiciously selected in a time-varying fashion so as to cause a negligible distortion in a standard receiver for analog FM. Simulations based on conservative nonoptimized signal design indicate that data rates up to about 130 kb/s inside the 200-kHz FM channel are achievable for acoustic test signals. We present a number of numerical examples where the average digital data use up to 50% of the 200-kHz power spectrum with digital signal power levels 25-35 dB below the analog signal. Due to the resulting variable-rate digital transmission, a control channel is required. A method of precanceling with multiple orthogonal direct-sequence spread-spectrum schemes is also presented     

Digitized speech transmission at VHF using existing FM mobile radios

An extensive study of digitized speech transmission over existing VHF FM mobile radio sets is presented. Objective criteria expressed in terms of bit error rate (BER) are determined for the performance evaluation of analog FM radios when transmitting digital voice. These criteria are used to define the technical characteristics such as the receiver sensitivity, co-channel rejection, and adjacent channel selectivity in the case of digital FM transmission. The concept of performance measurements of existing sets in the digital mode of operation is suggested. The results of performance measurements reported here include the following: 1) radiated FM spectra, 2) bit error rate curves, 3) receiver sensitivity, 4) BER performance in the presence of analog or digital FM interference on the same or adjacent channel, 5) signal plus noise plus distortion to noise plus distortion ratio (SINAD) performance in the presence of co-channel or adjacent channel interference for both types of interfering signals, 6) co-channel rejection and adjacent channel selectivity for various combinations of the wanted and interfering signals, 7) receiver selectivity for both modes of operation, and 8) adjacent-signal selectivity for various combinations of the desired and undesired signals. Optimum values of design parameters of digital voice radio system are deduced from the obtained results. Finally, the implications of digitized speech transmission on the technical characteristics, operating range, channel reuse distance, and adjacent channel interference performance of analog FM radios are discussed.     

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     

Decision feedback sequence estimation for continuous phasemodulation on a linear multipath channel

An approach to reduced-complexity detection of partial response continuous phase modulation (CPM) on a linear multipath channel is presented. The method, referred to as decision feedback sequence estimation (DFSE), is based on a conventional Viterbi algorithm (VA) using a reduced-state trellis combined with decision feedback (DF). By varying the number of states in the VA, the receiver structure can be changed gradually from a DF receiver to the optimal maximum-likelihood sequence estimator (MLSE). In this way different tradeoffs between performance and complexity can be obtained. Results on the receiver performance, based on minimum distance calculations and bit error rate simulations, are given for Gaussian minimum-shift keying modulation on typical mobile radio channels. It is shown that for channels with a long memory, a significant complexity reduction can be achieved at the cost of a moderate degradation in performance     

Precoded Modulo-Precanceling Systems for Simulcasting Analog FM and Digital Data

We present techniques for simulcasting low-power digital data and analog FM over fading channels. Our methods have strong connections to what are referred to as dirty paper coding techniques in that a low-power digital data signal is modulo-added to the host analog FM signal. Due to the low power levels of the digital data and the suppression capability of analog FM, a standard analog FM receiver can be used to reliably recover the analog audio signal. We develop digital receivers that work reliably over fading channels given imperfect channel state information. Spreadresponse precoding is exploited to not only provide rate-1 temporal diversity benefits, but also to simplify the design of the modulo-precanceler and the receiver. Our theoretical and simulation-based performance analysis of the digital receivers and our MSE-based analog FM distortion analysis suggest that precoded modulo-precanceling schemes provide substantially higher data rates than existing precanceling strategies at the cost of transmission delay and transmitter and receiver complexity. The systems we present are also readily compatible with conventional channel coding techniques as well as with analog FM postcancelers, which can improve the digital receiver bit-error-rate performance at the expense of receiver complexity.     

Equalization of microwave digital radio channels with a hybridacousto-optic and digital processor

Digital radio transmission systems use complex modulation schemes that require powerful signal processing techniques to correct channel distortions and to minimize bit-error rates (BERs). Combined analog and digital processors are investigated for minimizing the mean square error (MSE) of the radio receiver. The analog filters are implemented using acousto-optic (AO) processing since rapidly adaptable, inverse channel filters can be produced for either minimum or nonminimum phase channels. A specific architecture is identified and a laboratory system is tested to verify the ability of the processor to track and correct time-varying channels. Computer simulations are used to show that hybrid analog and digital equalization allows an increase in the modulation capacity of radio, relative to all digital equalization, while maintaining similar equipment signatures     

Postcanceling techniques for simultaneous broadcasting of analog FM and digital data

We consider simultaneous broadcasting of low-power digital data and analog FM and present reliable receivers for the digital data. Due to the relatively low power level of the digital data and the interference suppression capability of analog FM, standard analog FM receivers can reliably recover the analog FM audio signal. To recover the digital data, an extended Kalman filter front end is developed that exploits the capture capability of analog FM to reconstruct and postcancel the analog FM component from the received composite signal. Simulations are conducted with artificial analog FM signals, suggesting that postcanceling schemes can provide higher data rates than their precanceling counterparts, at a lower transmission delay penalty but higher digital receiver complexity. For analog FM to digital signal power level ratios in the range of 30-50 dB, the postcanceler digital signal recovery appears fairly robust, providing digital signal-to-noise ratios of 2-7.5 dB. The corresponding uncoded bit error rates strongly depend on the power level difference between the host analog FM and the digital data signal. In particular, at 260 kb/s and E/sub b//N/sub o/=10 dB they range between 1% to about 15%, and can be reduced to acceptable levels using standard channel coding techniques.     

创新的中国数字调频广播技术方案——CDRadio与HD Radio的简介与比较

数字化正对中国广播电视行业产生着深刻的影响,传统的FM音频广播也面临着数字化改造.在DAB,HD Radio,DRM及CDRadio几种先进数字音频广播技术方案中重点比较了HD Radio与CDRadio.中国数字调频广播技术方案CDRadio实现了真正的带内数模混叠同播,并通过使用非规则频谱分配、LDPC纠错编码、时...     

Mean-square crosstalk approach to CPM

The maximum-likelihood sequence estimator (MLSE) for continuous phase modulation (CPM) signals in an additive white Gaussian noise (AWGN) channel is a very efficient method of detection. This paper describes an extension of a relatively simple crosstalk approach for the performance analysis of linear quadrature receivers with cochannel interference (CCI) and adjacent channel interference (ACI) present to the MLSE receiver. Many CPM signals are analyzed, including those using new baseband modulating pulses. One of the new schemes allows an ACI signal to be 62 dB greater than the desired user signal at a frequency separation of one-and-a-half times the bit rate, with just a 2-dB degradation in required E_b/N₀     

Partial response continuous phase modulation and DPCM coding for speech transmission in cellular mobile radio systems

The performance of a speech transmission scheme with application to cellular digital mobile radio systems is considered. The source coder is embedded differential pulse code modulation (DPCM) and the modulation schemes belong to the class of partial response continuous phase modulation (CPM). Both quantizing noise and transmission errors contribute to the overall mean square error. The performance measure is the audio signal-to-noise ratio (SNR). It is seen that in a fading environment space diversity is very effective in bringing down the threshold of channel SNR to maintain the required audio SNR. The number of channels the system can support is evaluated under various conditions.     

Digital audio broadcasting in the FM band by means of contiguous band insertion and precanceling techniques

A low-complexity method of linearly combining an analog frequency-modulation (FM) signal with a low-level multitone signal for simultaneous broadcasting of digital data and analog FM is proposed and evaluated. The analog FM signal can be recovered with a standard FM receiver. The digital multitone signal is added symmetrically at the band edges of the 200-kHz FM signal in contiguous bands with variable width. To maximize the data rate, the digital bandwidth is maximized subject to a maximum allowed distortion of the demodulated analog FM signal. The impact of the analog FM signal on, digital reception is cancelled by means of transmitter precanceling techniques. Conditions for precanceling techniques for multipath channels are formulated. The main advantage of the contiguous band insertion method is that the rate of the control channel is considerably lower compared to previously proposed methods of carrier insertion without significant reduction in data rate. We present a number of numerical results for acoustical signals.     

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。     

Transmission of block turbo coded digital audio over FM-SCA

In this paper, an efficient coded scheme for transmitting digital audio over the existing FM channel, by multiplexing it with the baseband FM signal, is described. Transmission of multiplexed signals in the FM baseband called FM-SCA (subsidiary communications authorization) has been previously used for low quality analog content and some low rate digital content. The investigated scheme opens up the possibility of achieving CD quality audio over FM-SCA by enabling high bitrate transmission using MPEG-I layer 3 and MPEG-AAC audio coding for the digital audio. These schemes provide CD quality audio at or below 128 kbps, MPEG-AAC being able to do so at rates as low as 96 kbps. Block turbo codes (BTC), which offer near Shannon's limit performance with relatively low hardware complexity requirements, provide the error protection. Block turbo codes have been shown to be particularly effective for high coding rates. The system uses OFDM in conjunction with 8PSK/16PSK to modulate the digital bitstream and fit it in the 44 kHz (54 to 98 kHz) band available in the FM baseband. Simulation results show an optimal system configuration for digital audio transmission in FM-SCA.     

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     

Soft-output demodulator in space-time-coded continuous phase modulation

Space-time coding has shown great promise for digital transmission in wireless communication links, especially when the channel response is known at the receiver. Space time coding combined with continuous phase modulation (CPM) can offer better tradeoffs in bandwidth and power efficiency. Because of the memory inherent in CPM, channel estimation is often harder than for linear modulations. We present an adaptive soft algorithm that performs joint channel estimation and data detection for space-time CPM systems. Properly designed pilot symbols are inserted at the very beginning to give good initial estimates of the channels. This soft receiver is further applied to the interleaved space-time CPM system to yield better performance with moderate complexity through iterative processing. Simulation results show that the receiver can often achieve near-coherent performance in quasistatic fading as well as in time-varying fading.     

一种无线语音传输系统设计

利用锁相环技术,通过调频方式,实现音频信号的调制、解调,设计一种可以选择使用耳机或扬声器的无线语音传输系统.系统利用电路实现语音的通信和信息转发,可实现对语音实时无失真传输,系统实现对音频信号的无线通信的功能.     

软件无线电电台的计算机仿真

分析了软件无线电的分层结构,设计了一种软件无线电电台模型,其射频段参数、信道模型、调制方式和信源编码方式等可编程控制。建立了一种软件无线电电台的计算机仿真平台,进行了计算机仿真,验证了软件无线电的一些重要思想和技术。针对AM,SSB,FM,OOK,PSK,QPSK,MSK,FSK,16QAM九种信号,仿真结果表明,在载波频率、调制方式、信道模型等可以根据需要设置的环境下,电台的收发都能达到预期效果。当接收信号的信噪比为20 dB,AM与SSB调制度为0.8,FM的调频指数为3、数字调制的传速率都为2000 b/s时,载波频率的平均检测误差不大于2%,平均调制类型识别正确率不低于96%,模拟调制的平均失真率小于3%,数字调制的平均误码率小于10-5。     

A reduced-state Viterbi receiver with prefiltering for the mobile radio channel

This paper presents a possible receiver structure for high bit rate transmission on the mobile radio channel. The receiver is based on a reduced-state Viterbi algorithm combined with decision feedback, known as Decision Feedback Sequence Estimation. A tdma concept using a training sequence for channel estimation and continuous phase modulation is assumed. For channel estimation a least squares method is studied, and an adaptive prefilter is used to improve the receiver performance on minimum phase channels. Estimates on the computational complexity of the receiver and simulation results showing the performance on a typical urban (tu) channel are presented. The results show that the prefilter improves the performance of the receiver significantly when a small number of states is used in the Viterbi algorithm. However, the computational complexity for 2 Mbit/s continuous transmission is very high.     

OFDM混合调制信号在认知网络中的应用

对OFDM混合调制信号进行了谱相关分析,并将其应用到认知网络中。在采用OFDM混合调制信号传输的认知网络中,利用信号的循环平稳特性可以快速识别出信道空闲、认知信号传输、主用户信号传输和两种信号混叠这四种状态,为认知网络中认知信号的快速可靠接入和退出提供了保障。进一步利用混合调制的位置来标记不同的认知网络,认知节点可以区分不同认知网络信号。在提出的认知网络应用模型中,传输和接收设备较简单,而且不需专门的信道来传输控制命令,节约了带宽。仿真结果显示,在认知网络中,该模型可以提高频谱利用率,使通信系统的性能得到改善。     

Complementary punctured-pair convolutional codes for digital audio broadcasting

Hybrid in-band on-channel (IBOC) broadcasting systems for digital audio radio have the capability of simultaneously transmitting analog FM and digital audio of CD-like quality. Due to fading and interference in the already-crowded FM band, the signal design for the hybrid IBOC system is very challenging. It has been proposed to use a method of double sideband transmission where the digital information is transmitted by means of orthogonal frequency-division multiplexing (OFDM) on both sides of the analog host FM and where the digital information can be recovered when one sideband is partially or even totally lost. This leads to an interesting channel coding problem, where we search for optimal pairs of high-rate codes that form good combined low-rate codes, which are better than classic code combining techniques. Furthermore, we also search for rate-compatible punctured convolutional codes which can be used for two-level unequal error protection (UEP) of digital audio. Since some of the tones in the multitone modem (OFDM) are more exposed to interference than others, optimal assignments of convolutional code bits to tones depending on their spectral position are also found. A large number of new codes with memory 6 and 8 are presented both for equal error protection and UEP.