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     

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.     

The Effects of Degraded Audio on Memory for Speech Passages

As the US migrates to digital radio, broadcasters are making important decisions about the transmission rates they use on primary and secondary audio channels. One way to evaluate coders at different bit rates is to elicit consumers' opinions of the audio quality. However, it is evident from several lines of research that consumers' preferences do not always match their ability to process information efficiency. Therefore when evaluating coders at low bit-rates it is important to measure consumer's processing efficiency as well as examining their preferences. With regard to audio, past research has shown that degraded speech, including speech masked with noise and synthetic speech, impairs individuals' ability to recall content. The current research explores whether speech coded at very low bit rates (i.e., 9 kbps and 24 kbps) also impairs consumers' memory for text passages. Concurrently, we examine people's subjective ratings to see whether these ratings correlated with our objective recall measure. Results suggest that participants' recall of information dropped significantly at 9 kbps, a finding which has important implications for broadcasters. Participants' audio quality ratings did not correlate with their ability to remember story details, supporting the notion that merely asking consumers to rate the quality of a signal may not provide a complete picture of how impaired audio will affect their behavior. However, interestingly, their professed level of interest, comprehension, and enjoyment with the story did correlate with their memory scores, making these questions better candidates for subjective tests of very low bit rate coders.      

A Power Series Expansion Approach to Evaluate Modulation Transfer Noise Effects from Digital Carriers to FDM/FM Carriers in Memoryless Nonlinear Devices

This paper presents a power series expansion approach for evaluating modulation transfer noise effects fromLdigital carriers (continuous or bursty) toKFDM/FM carriers in a common memoryless nonlinear amplifier. The method allows for a complete characterization and assessment of this type of transmission impairment and its effects in satellite communications systems. The baseband modulation transfer noise in the FM carriers caused by the random envelope fluctuations due to filtering of PSK pulses and/or by the on-off bursting of the digital carriers can be calculated. Illustrative numerical results indicate that the mixed PSK-FM mode of operation in satellite transponders should be employed with great care since a number of 4 kHz channels in the basebands of the FM carriers may receive prohibitive interference.     

Performance evaluation study of signal design and equalization options for high-bit-rate digital subscriber line transceivers

This paper analyses different equalization, coding and signal constellation alternatives for the proposed high-bit-rate (800 kb/s) digital subscriber loop transmission interface. The simulation results presented in the paper demonstrate that bit rates above 1000 kb/s at acceptably low bit-error rates (≤ 10^?7) are feasible, if baseband transceivers with appropriate equalization and simple trellis coding are used. It is also shown that baseband transmission and Tomlinson precoding provide a significant performance advantage over bandpass transmission and decision-feedback equalization, respectively.     

Concatenated coding schemes and interleaving techniques over Rayleigh fading ionospheric channels

In this paper the performance of a concatenated coding scheme is evaluated over a slow Rayleigh fading HF ionospheric link with additive white Gaussian noise (AWGN). Well‐known Ungerboeck TCM techniques onto an 8‐PSK signal set are used as inner codes and Reed–Solomon block codes as outer codes. The coded/modulated signal is further differentially encoded before transmission to combat random phase changes. Block interleaving techniques are necessary to randomise long bursts of errors caused by the fading channel. The performance of the proposed concatenated coding system is investigated for various Doppler spreads. Significant coding gains are achieved over uncoded, diversity or other conventionally coded systems with a small bandwidth expansion. Finally the interesting effects of interleaving on the behaviour of the proposed systems are analysed. Copyright © 2001 John Wiley & Sons, Ltd.     

A novel turbo coding scheme for satellite ATM using Reed-Muller codes

Block turbo codes with trellis-based decoding are proposed for use in cell-based satellite communication. Shortened Reed-Muller (RM) codes are used as the component codes because their minimal trellis is known. Simulation results for RM turbo codes and shortened RM turbo codes are presented over additive white Gaussian noise and Rayleigh fading channels. The performance of the shortened codes with different shortening patterns are shown. In some cases, the codes have the unequal error protection property, useful in asynchronous transfer mode cell formatting. In order to test the suitability of the proposed coding scheme from a practical point of view, the effect of channel impairments, including channel signal-to-noise ratio mismatch and carrier phase offset, are investigated.     

基于小波基优化和动态比特分配的数字音频压缩编码方案

提出了一种新颖的基于自适应小波基优化选择和心理声学模型相结合的数字音频信号的透明质量编码方法，保证固定失真水平上使每帧信号的变换系数的动态分配的比特数最少，并且利用动态码本的方法来消除音频信号的统计冗余，进一步压缩比特率，对于抽样率为４４．１ｋＨｚ每样值用１６比特线性码表示的光盘单声道音乐信号可以压缩到６４ｋＢＰＳ左右。     

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.     

AMR-WB+技术性能分析和测试

第三代移动通信系统提供的语音短信、流媒体和数字广播服务有着广阔的商业空间和发展潜力,所以一种高效低码率的音频编码方法对移动通信就有着极其重要的意义.3GPP已经把AHR-WB 格式作为其3G流媒体音频部分的编码标准.在低编码速率范围内(JA10Kbps到24Kbps),AHR-WB 编解码格式对音频的处理表现出独特的性能.本文对AMR-WB 进行较深入的研究,并分析和测试AHR-WB 算法的改进带来的音质改善.     

An introduction to super audio CD and DVD-Audio

Highlights the latest developments in consumer audio and specifically in DVD-Audio and SACD. The DVD-Audio specification allows for up to 24-b PCM data and uses the Meridian lossless packing (MLP) algorithm to provide up to six channels of high-quality, multichannel audio at sampling rates of up to 96 kHz for six channels or 192 kHz for two channels. Super-audio CD (SACD), introduced in March 1999, integrates a variety of new technologies, such as the hybrid disc, direct stream digital (DSD), and direct stream transfer coding.     

Evaluating the performance of turbo codes and turbo-codedmodulation in a DS-CDMA environment

Turbo codes have received great attention due to their outstanding performance. Unfortunately, a high performance is associated with large transmission delays, prohibiting an application for speech transmission. Hence, the aim of this paper is the comparison of turbo codes employing short interleavers with convolutional codes in terms of bit error rate performance and computational effort. Additionally, a pragmatic approach of bandwidth-efficient turbo-coded modulation is considered. Analyzing the structure of the transmitter and receiver, interesting results are presented concerning the design of the mapper. Furthermore, a new very simple soft-output demodulation algorithm is derived. In order to compare turbo codes with convolutional codes under realistic conditions, both are embedded in a direct sequence (DS) code division multiple access system. Besides this comparison, a compromise between a high coding gain (low code rate) and high direct-sequence spreading is worked out, including the consideration of the turbo-coded modulation scheme. Simulation results indicate that turbo codes with small block interleavers do not outperform conventional convolutional codes. Furthermore, it is shown that for coherent demodulation, low code rates and low DS spreading is superior to high code rates and high DS spreading     

An efficient optimisation scheme for scalable surveillance centric video communications

State-of-the-art coders have been optimised over years according to the needs of the broadcasting industry. There are however key applications of coding technology whose challenges and requirements substantially differ from broadcasting. One of these key applications is surveillance. In this paper an efficient approach for surveillance centric joint source and channel coding is proposed. Contrasting conventional coders, the proposed system has been developed according to the requirements of surveillance application scenarios. It aims at achieving bit-rate optimisation and adaptation of surveillance videos for storing and transmission purposes. In the proposed approach the encoder communicates with a video content analysis (VCA) module that detects events of interests in video captured by CCTV. Bit-rate optimisation and adaptation is achieved by exploiting the scalability properties of the employed codec. Temporal segments containing events relevant to surveillance application are encoded using high spatio-temporal resolution and quality while the portions irrelevant from the surveillance standpoint are encoded at low spatio-temporal resolution and/or quality. Furthermore, the approach jointly optimises the bit allocation between the wavelet-based scalable video coder and forward error correction codes. The forward error correction code is based on the product code constituting of LDPC codes and turbo codes. Turbo codes show good performance at high error rates region but LDPC outperforms turbo codes at low error rates. Therefore, the concatenation of LDPC and TC enhances the performance at both low and high signal-to-noise (SNR) ratios. The proposed approach minimises the distortion of reconstructed video, subject to constraint on the overall transmission bit-rate budget. Experimental results clearly demonstrate the efficiency and suitability of the proposed approach in surveillance applications.     

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     

An efficient baseband modem technique for data transmission overanalog FM links

In mobile radio where data are transmitted over existing analog FM systems, the receive bandpass bandwidth, which is adapted to the analog speech transmission, is larger than would be required by data transmission. This results in poor error performance. A novel baseband modem technique which drastically improves the error performance is proposed and analyzed. A smearing filter is used to convert the compound baseband noise at the limiter discriminator output to approximately Gaussian noise. This optimizes the performance at large carrier-to-noise ratios (CNRs). A baseband click detection and elimination scheme, which improves the performance at small CNR is proposed. Simulation results show that this system requires 3.8 dB less CNR than the conventional digital FM system to achieve a bit error probability of 10^-4. It is concluded that the technique is attractive for data transmission over analog FM links     

Transmission scheme of high-capacity FM multiplex broadcastingsystem

FM multiplex broadcasting is a system for providing additional text and graphics, while maintaining compatibility with existing stereo sound broadcasting. The digital signals are multiplexed in a higher frequency band than baseband FM stereo signals. This paper describes a modulation method and an error correction method for a new high-capacity FM multiplex broadcasting system called DARC (Data Radio Channel), which has a bit rate of 16 kbit/s. Simulation results show that stereo sound signals interfere with a multiplexing signal under multipath conditions. LMSK (level controlled MSK) is proposed as a modulation scheme to ensure good transmission quality. It is shown that an error correction scheme using a product code of (272190) codes has a good performance for mobile reception. Field tests on the DARC for mobile reception are conducted in the service area of the NHK Tokyo FM station. These show that the correct reception rate can be obtained at more than 80% when transmitting information of 6 kbytes     

Speech and data communications over 942 MHz TAB and TTIB single sideband mobile radio systems incorporating feed-forward signal regeneration

The transmission of speech and data over 942 MHz pilot tone single sideband (SSB) mobile radio links is the main concern of this paper. It has been found that the use of feedforward signal regeneration enables a speech quality to be obtained in the Rayleigh fading environment which is superior to that achieved by a 25 kHz Advanced Mobile Phone Service (AMPS) type FM system and markedly superior to that obtained with a 12.5 kHz FM system. A new optimized form of SSB, phase-locked transparent tone-in-band (TTIB), is shown to be capable of achieving coherent data transmission such as M-ary phase shift keying (PSK) in the presence of Rayleigh fading without the usual "high-level" irreducible error rates. The signal processing described has wide application from line to satellite communications.     

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.     

Performance of multilevel‐turbo codes with blind/non‐blind equalization over WSSUS multipath channels

In this paper, in order to improve error performance, we introduce a new type of turbo codes, called ‘multilevel‐turbo codes (ML‐TC)’ and we evaluate their performance over wide‐sense stationary uncorrelated scattering (WSSUS) multipath channels. The basic idea of ML‐TC scheme is to partition a signal set into several levels and to encode each level separately by a proper component of the turbo encoder. In the considered structure, the parallel input data sequences are encoded by our multilevel scheme and mapped to any modulation type such as MPSK, MQAM, etc. Since WSSUS channels are very severe fading environments, it is needed to pass the received noisy signals through non‐blind or blind equalizers before turbo decoders. In ML‐TC schemes, noisy WSSUS corrupted signal sequence is first processed in equalizer block, then fed into the first level of turbo decoder and the first sequence is estimated from this first Turbo decoder. Subsequently, the other following input sequences of the frame are computed by using the estimated input bit streams of previous levels. Here, as a ML‐TC example, 4PSK 2 level‐turbo codes (2L‐TC) is chosen and its error performance is evaluated in WSSUS channel modelled by COST 207 (Cooperation in the field of Science & Technology, Project #207). It is shown that 2L‐TC signals with equalizer blocks exhibit considerable performance gains even at lower SNR values compared to 8PSK‐turbo trellis coded modulation (TTCM). The simulation results of the proposed scheme have up to 5.5 dB coding gain compared to 8PSK‐TTCM for all cases. It is interesting that after a constant SNR value, 2L‐TC with blind equalizer has better error performance than non‐blind filtered schemes. We conclude that our proposed scheme has promising results compared to classical schemes for all SNR values in WSSUS channels. Copyright © 2005 John Wiley & Sons, Ltd.