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     

MF digital AM station antennas

It is very important having the proper antenna, specially in the MF AM band where achieving the necessary bandwidth in the antenna impedance is a difficult task. This problem is critical in the lower frequencies were the antenna matching to the transmission line generally is very sharp and the best match is obtained only at the carrier frequency with an appreciable power reflection in the lateral frequency bands. This problem is not very important in the classical AM transmissions were the maximum transmitted power is located at the carrier frequency and only a fraction in the upper and lower lateral bands. Of course this produces some distortion in the AM transmission but in this case the quality of the audio is not really of high fidelity, like in an FM transmission, due to a lot of factors, one of them the lack of the full audio spectrum. This problem can be corrected with high fidelity audio transmitters and specially with digital transmission in order to achieve CD quality audio and here the transmitting antenna plays an important role. In this paper MF AM antenna systems are analyzed not only from the input impedance point of view, but with consideration of all the factors in order to determine the best system in bandwidth and radiation properties in different parts of the standard AM band. Cylindrical, type A, and Cantilever classical monopoles and the modern dipole type antenna systems are compared, to provide the criteria for choosing an optimum antenna for the future digital AM service. Examples of measured field strength as a function of distance in a flat region are presented in order to show the interesting MF AM possibilities for a digital service     

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     

Design and test of a spectrally efficient land mobile communications system using LPC speech

A communication system was built and tested to operate in the land mobile VHF band (150-174 MHz) at a channel separation of only 6 kHz. The audio source was digitally encoded at 2.4 kbits/s using linear predictive coding (LPC). The speech data stream was transmitted by frequency shift keying (FSK) which allowed the use of class-C transmitters and discriminator detection in the receiver. Baseband filtering of the NRZ data resulted in a narrow transmitter spectrum. The receiver had a 3 dB bandwidth of 2.4 kHz which allowed data transmission with minimal intersymbol interference and frequency offset degradation. A 58 percent eye opening was found. Bit error rate (BER) performance was measured with simulated Rayleigh fading at typical 150 MHz rates. Additional tests included capture, ignition noise susceptibility, adjacent channel protection, degradation from frequency offset, and bit error effects upon speech quality. A field test was conducted to compare the speech quality of the digital radio to that of a conventional 5 kHz deviation FM mobile radio.     

Digital Transmission of Commentary-Grade (7 KHz) Audio at 56 or 64 Kbits/s

A two-band coding system has been constructed for the purpose of providing commentary grade (7 kHz bandwidth) speech or music transmission at 56 or 64 kbits/s. The lower band, 0-to-3650 Hz, is coded with 4 bit ADPCM and the upper band, 3600-to-6800 Hz, is coded with 3 bit or 4 bit/sample APCM. The quality of the coded signal makes the method useful for news and sports broadcasts, and possibly for AM remote music broadcasting. The audio sounds better than that produced by two conventional alternatives: 3200 Hz bandwidth with 8-bit/sample coding and 7000 Hz bandwidth with a single 4-bit/sample coder. The sample may be used in any place with access to a 56 kbit/s Dataphone Digital Service port or to other 56 or 64 kbit/s lines. The power consumption is approximately 12 W in the present form; it could be reduced by a factor of at least two by hardware optimization.     

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     

Efficient Generation of Scalable Transport Stream for High Quality Service in T‐DMB

We introduce an advanced terrestrial digital multimedia broadcasting (AT‐DMB) system that overcomes the limitation of data transmission rates of T‐DMB by doubling it with the same frequency bandwidth. In this letter, we propose an efficient algorithm which generates a scalable transport stream in AT‐DMB by multiplexing certain types of elementary streams encoded using scalable video coding and an MPEG‐surround audio coder for high‐quality multimedia services.     

Design and Test of a Spectrally Efficient Land Mobile Communications System Using LPC Speech

A communication system was built and tested to operate in the land mobile VHF band (150-174 MHz) at a channel separation of only 6 kHz. The audio source was digitally encoded at 2.4 kbits/s using linear predictive coding (LPC). The speech data stream was transmitted by frequency shift keying (FSK) which allowed the use of class-C transmitters and discriminator detection in the receiver. Baseband filtering of the NRZ data resulted in a narrow transmitter spectrum. The receiver had a 3 dB bandwidth of 2.4 kHz which allowed data transmission with minimal intersymbol interference and frequency offset degradation. A 58 percent eye opening was found. Bit error rate (BER) performance was measured with simulated Rayleigh fading at typical 150 MHz rates. Additional tests included capture, ignition noise susceptibility, adjacent channel protection, degradation from frequency offset, and bit error effects upon speech quality. A field test was conducted to compare the speech quality of the digital radio to that of a conventional 5 kHz deviation FM mobile radio.     

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.     

Analysis of new methods for broadcasting digital data to mobileterminals over an FM-channel

Two new methods using an FM-radio channel for transmission of digital data to mobile terminals are examined: 1. A modification of the radio data system (RDS). In RDS, additional digital information is multiplexed with a stereo sound signal. A new system is suggested where the data signal can be multiplexed with a mono audio signal. This causes extension to the bandwidth available for the data signal, and therefore the RDS bitrate can be increased. Error calculations are performed both for the original RDS system and for the new system. 2. Orthogonal frequency division multiplexing (OFDM). OFDM is used in the digital audio broadcasting system (DAB), which is designed to transmit digital audio in the FM band. In OFDM a signal is divided over a large number of 2- or 4-PSK modulated orthogonal subcarriers. The subcarriers of 6 different programmes are multiplexed in one beam to reduce the effects of frequency selectivity of the transmission channel. A new system based on OFDM is proposed, in which the carriers of each programme are transmitted in one FM-channel with a bandwidth of 200 kHz instead of multiplexed with the carriers of other programmes. Error calculations are performed for the subcarriers used in the OFDM modulation method     

基于Simulink的窄带OFDM系统设计与仿真

为有效提高短波通信系统的数据传输速率,可采用正交频分复用（OFDM）技术。OFDM是一种特殊的多载波数字调制方案,各子载波之间的正交性使得频谱可重叠从而提高了频带的利用率,同时能有效对抗频率选择性衰落或窄带干扰。采用Simulink建模对窄带OFDM系统进行设计并仿真,仿真结果表明,在3kHz的带宽内,短波通信系统的数据传输速率能够达到12800b／s,而且具有良好的误码性能。     

Digital audio broadcasting in the FM band based on continuous phase modulation

A method for broadcasting digital audio signals simultaneously with existing analog frequency modulation radio (88-108 MHz) in adjacent channels is presented. The digital transmission is based on continuous phase modulation (CPM) and a proper reduced-state sequence estimator. With the proposed method, the power level and the symbol rate of the transmitter signal is determined in a manner that the interference the CPM signal poses for the analog FM signal in adjacent channels remains below a level according to the radio frequency emission mask defined by international rules. Due to the multipath propagation of the transmitted signal, the transmission behavior of the radio channel is determined by high dispersion up to 85 /spl mu/s. With the selected bit rate, the receiver has to cope with a channel memory of up to 17 bits. Since Viterbi detection is not feasible due to the number of channel states, detection is performed by a reduced-state sequence estimator that is able to eliminate the complete channel interference by decision feedback. Simulation results show that the detector almost achieves the detection quality of the optimum receiver. CPM achieves data rates of up to 200 kb/s inside a 200 kHz FM channel, which is sufficient for transmission of digital compressed audio signals at compact disc quality. The encouraging results of field tests will be published in another paper.     

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.     

基于软件无线电的中短波接收机设计

目前广播系统正由模拟体制向数字化体制过渡,数字版权管理（DRM,Digital Radio Mondiale）系统就是针对30 MHz以下频段的数字广播标准。作为软件无线电的核心技术之一,数字接收机技术得到了越来越普遍的应用。介绍了一种基于软件无线电的中短波接收机的硬件系统结构,这种软件无线电接收机结构可以同时解码音频和数据流,兼容目前的模拟调幅广播和数字调幅广播标准。重点讨论了电调滤波器的设计,最后使用EDA软件进行仿真,并给出了仿真结果。     

A Scalable Three Bit-rates 8–14.1-24 kbit/s Audio Coder

This paper presents a scalable three bit- rates (8, 14.1 and 24 kbit/s) coder. For the two embedded lowest ones, operating in the telephone bandwidth, celp coding techniques are used. For the highest rate, that both improves narrowband quality and extends the band to [50-7000Hz], transform coding techniques are used. The main applications deal with transmission over network with no guaranteed QoS.     

EIA/NRSC DAR systems subjective tests. II. Transmission impairments

For pt.I see ibid., vol.43, no.3, p.261-7 (1997). This paper describes the methods, results and conclusions of a series of subjective tests which were performed at the Communications Research Centre, Ottawa, Canada, to assess the audio quality of digital audio radio (DAR) systems in the presence of transmission errors. Testing has been performed in a laboratory environment on the hardware implementation of nine DAR systems. One of the systems operated in the AM broadcast band, five in the FM broadcast band, two in the L-band and one in the S-band. Testing was done in the presence of transmission errors generated by additive white gaussian noise, co-channel interference and five different multipath mobile channels. In addition, one system was tested with an interfering signal in the lower first adjacent DAR channel. The test results are presented and discussed, for each type of transmission impairments, in terms of (a) the E_b/N₀ ratios (D/U ratios for co-channel and adjacent channel interference) at the threshold of audibility (TOA) of transmission errors and at the point where transmission errors are so important that the audio quality is unacceptable (point of failure or POF) and (b) the failure margin which is a measure of how quickly DAR systems fail when the received signal power is reduced. A new parameter, labeled C_SP/N₀, is proposed to quantify the power efficiency of the overall DAR systems (i.e. source coding plus channel coding and modulation subsystems). The spectral efficiency of DAR systems is also presented and discussed     

数字调制系统码间干扰分析

通信系统中引起误码的主要因素是噪声和码间干扰。主要针对频带传输系统包括多进制调制系统如何能消除码间干扰问题进行了分析，提出了无码间干扰传输系统的传输函数应满足的条件、最高码元传输速率与系统带宽之间的关系以及不同调制系统的频带利用率。