Speech and channel coding for digital land-mobile radio

The joint development of a medium bit-rate speech coder along with an effective channel coding technique to provide a robust, spectrally efficient, high-quality mobile communication system is described. A subband coder operating at 12 kb/s is used; in the absence of channel errors, it provides speech quality comparable to current analog land-mobile radio systems. The coder design incorporates a unique coding of the side information to facilitate the use of forward-error-correction coding without the need to code the entire bit stream. The use of excessive overhead for redundancy is avoided while the harsh effects of frequent channels are mitigated. These techniques have been used in an experimental FDMA (frequency-division multiple access) digital land-mobile radio system. The combined speech and channel coder operates at 15 kb/s and provides intelligible speech at fading channel error rates up to 8%     

Wideband DS-CDMA for next-generation mobile communications systems

Wideband wireless access based on direct sequence code division multiple access aimed at third-generation mobile communications systems is reviewed. W-CDMA is designed to flexibly offer wideband services which cannot be provided by present cellular systems, with various data rates as high as 2 Mb/s. The important concept of W-CDMA is the introduction of intercell asynchronous operation and the pilot channel associated with individual data channels. Intercell asynchronous operation facilitates continuous system deployment from outdoors to indoors. Other technical features of W-CDMA include fast cell search under intercell asynchronous operation, fast transmit power control, coherent spreading code tracking, a coherent RAKE receiver, orthogonal multispreading factor forward link, and variable-rate transmission with blind rate detection. The introduction of the data-channel-associated pilot channel allows W-CDMA to support interference cancellation and adaptive antenna array techniques that can significantly increase the link capacity and coverage. This article presents the radio link performance evaluated by computer simulation. Field experiment radio link performance results are also presented     

Advanced Cordless Telecommunications Service

Present analog cordless telephones suffer from lack of privacy, limited facilities and can support only a relatively low density of users. This restricts their benefits and market opportunities. Hence the need for an advanced cordless telecommunications service (ACTS). Consideration is given here to the needs of this service in terms of requirements, demand, system aspects, spectrum needs, propagation, and performance. This leads to a proposed system design for the service which is already under consideration within Europe. It uses 32 kbit/s speech encoding and time-division duplex operation to achieve both-way speech communication on a single 100 KHz-wide radio channel. A pool of radio channels are available to choose from at both base and handset in order to minimize overall interference and maximize system capacity. The addition of another aerial at only the base unit can provide the full spatial diversity improvement to both directions of transmission. The service is due for introduction in the United Kingdom in 1987.     

Characterizing the subjective performance of the ITU-T 8 kb/sspeech coding algorithm-ITU-T G.729

Three listening-only experiments were conducted to characterize the subjective performance (i.e., speech quality) of 8 kb/s G.729. These experiments evaluated the quality of coded speech under a variety of conditions: (i) interworking with other international and regional speech coding standards; (ii) input speech that had been corrupted by environmental noise; (iii) operation over degraded transmission channels (including random bit errors and a simulated radio channel). The results of these experiments indicate that 8 kb/s G.729 meets the performance requirements that were established at the beginning of the standardization process     

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.     

High-quality coding of telephone speech and wideband audio

Digital speech technology is reviewed, with the emphasis on applications demanding high-quality reproduction of the speech signal. Examples of such applications are network telephony, ISDN terminals for audio teleconferencing, and systems for the storage of audio signals, which include the important subclass of wideband speech. Depending on the application, the bandwidth of input speech can vary from about 3 kHz to nearly 20 kHz. Coding for digital telephony at 4 and 8 kb/s, network quality coding at 16 kb/s, and coding for audio at 7 and 20 kHz are examined. Future directions in the field are discussed with respect to anticipated technology applications and the algorithms needed to support these technologies     

A generalized RAKE receiver for interference suppression

Currently, a global third-generation cellular system based on code-division multiple-access (CDMA) is being developed with a wider bandwidth than existing second-generation systems. The wider bandwidth provides increased multipath resolution in a time-dispersive channel, leading to higher frequency-selectivity. A generalized RAKE receiver for interference suppression and multipath mitigation is proposed. The receiver exploits the fact that time dispersion significantly distorts the interference spectrum from each base station in the downlink of a wideband CDMA system. Compared to the conventional RAKE receiver, this generalized RAKE receiver may have more fingers and different combining weights. The weights are derived from a maximum likelihood formulation, modeling the intracell interference as colored Gaussian noise. This low-complexity detector is especially useful for systems with orthogonal downlink spreading codes, as orthogonality between own cell signals cannot be maintained in a frequency-selective channel. The performance of the proposed receiver is quantified via analysis and simulation for different dispersive channels, including Rayleigh fading channels. Gains on the order of 1-3.5 dB are achieved, depending on the dispersiveness of the channel, with only a modest increase in the number of fingers. For a wideband CDMA (WCDMA) system and a realistic mobile radio channel, this translates to capacity gains of the order of 100%     

基于AMBE2000的新型无线同播系统

在无线多通道语音通信中,距离长短使各接收机接收信号的时间不一,同步输出需要精确的延时。介绍一种新型的基于AMBE2000和AD73311的高质量语音同步通信系统的设计与实现。该系统能够在2.0～2.4kb/s的低比特速率产生高话音质量,实现最小步长为1us的输出延时,达到多路同步传输效果。因此,该系统可以广泛应用于软件无线电,多模式电台及多路广播等场合。     

Analysis of TCP performance under joint rate and power adaptation in cellular WCDMA networks

To improve the spectral efficiency while meeting the radio link level quality of service requirements such as the bit-error-rate (BER) requirements for the different wireless services, transmission rate and power corresponding to the different mobile users can be dynamically varied in a cellular wideband code-division multiple-access (WCDMA) network depending on the variations in channel interference and fading conditions. This paper models and analyzes the performance of transmission control protocol (TCP) under joint rate and power adaptation with constrained BER requirements for downlink data transmission in a cellular variable spreading factor (VSF) WCDMA network. The aim of this multilayer modeling of the WCDMA radio interface is to better understand the interlayer protocol interactions and identify suitable transport and radio link layer mechanisms to improve TCP performance in a wide-area cellular WCDMA network.     

Soft iterative multisensor multiuser detection in coded dispersiveCDMA wireless channels

A soft iterative multisensor array receiver for coded multiuser wideband code-division multiple-access wireless uplink channels is proposed, such channels are typically both frequency- and time-selective. A new equivalent discrete-time synchronous representation is used to model asynchronous multiuser dispersive channels that employ, in general, random spreading. The proposed scheme suppresses multiuser interference over a wide range of user signal powers, by iteratively exchanging soft information between a minimum mean-square-error (MMSE) multiuser demodulator and a bank of single-user decoders, feeding back the outputs of the latter to aid in soft multiuser multisensor MMSE-RAKE demodulation and subtractive interference cancellation, in the second and subsequent iterations. It displays near-far resistance since it behaves like a successive interference subtracter across iterations. The array responses are obtained via a multipass estimation scheme that uses both (temporal) pilot symbols and soft estimates fed back from the decoders to effectively provide multisymbol pilot signals and thence successively refined estimates with increasing iterations, while seeking neither to rely on the array geometry nor to estimate the directions from which users' signals arrive at the receiver. Simulation studies indicate that this scheme performs close to the single-user case with a two-sensor receiver array, and perfect channel state information, after four iterations; alternatively, it allows significantly increased user capacity compared with conventional receivers, and suffers only a modest loss with estimated array responses     

Robust Cognitive‐Radio‐Based OFDM Architecture with Adaptive Traffic Allocation in Time and Frequency

Cognitive radio (CR) has been proposed as an effective technology for flexible use of the radio spectrum. The interference between primary users and CR users, however, becomes a critical problem when they are using adjacent frequency channels with different transmission power levels. In this paper, a robust CR orthogonal frequency division multiplexing (OFDM) architecture, which can effectively suppress interference to nearby primary users and overcome adjacent channel interference (ACI) to the CR user, is proposed. This new approach is characterized by adaptive data repetition for subcarriers under heavy ACI, and adaptive time spreading for subcarriers near the borders of the CR user's spectrum. The data repetition scheme provides extra power gain against the ACI coming from primary users. Time spreading guarantees an acceptable interference level to nearby primary users. By computer simulation, we demonstrate that, under a CR environment, the proposed CR OFDM architecture outperforms conventional OFDM systems in terms of throughput and BER performance.     

Downlink transmission of broadband OFCDM Systems-part II: effect of Doppler shift

The orthogonal frequency and code division multiplexing (OFCDM) system with 2-D spreading (time- and frequency-domain spreading) is becoming a promising candidate for future broadband wireless communication systems. OFCDM is more attractive than orthogonal frequency-division multiplexing (OFDM) both by introducing frequency-domain spreading for frequency diversity provision and time-domain spreading for flexible data rate provision. To provide high-speed mobile services, multicode transmission is employed in conjunction with OFCDM. In a Gaussian or flat-fading channel, multicode channels are orthogonal. However, in a realistic wireless channel, the orthogonality no longer maintains. Thus, multicode interference (MCI) is caused. This paper focuses on the investigation of the effect of Doppler shift on the downlink transmission of high-speed mobile OFCDM systems. A practical channel estimation algorithm based on a code-multiplexed pilot channel is employed to track the variations of fading channels. Hybrid MCI cancellation and minimum mean-square error (MMSE) detection proposed by the authors is employed as an efficient way to eliminate the MCI in the frequency domain. The system performance is analytically studied with imperfect channel estimation to show how it is affected by parameters such as the window size in the channel estimation, Doppler shift, the number of stages of the hybrid detection, the power ratio of pilot to data channels, spreading factor, and so on.     

MIMO-OFDM for wireless communications: signal detection with enhanced channel estimation

Multiple transmit and receive antennas can be used to form multiple-input multiple-output (MIMO) channels to increase the capacity by a factor of the minimum number of transmit and receive antennas. In this paper, orthogonal frequency division multiplexing (OFDM) for MIMO channels (MIMO-OFDM) is considered for wideband transmission to mitigate intersymbol interference and enhance system capacity. The MIMO-OFDM system uses two independent space-time codes for two sets of two transmit antennas. At the receiver, the independent space-time codes are decoded using prewhitening, followed by minimum-Euclidean-distance decoding based on successive interference cancellation. Computer simulation shows that for four-input and four-output systems transmitting data at 4 Mb/s over a 1.25 MHz channel, the required signal-to-noise ratios (SNRs) for 10% and 1% word error rates (WER) are 10.5 dB and 13.8 dB, respectively, when each codeword contains 500 information bits and the channel's Doppler frequency is 40 Hz (corresponding normalized frequency: 0.9%). Increasing the number of the receive antennas improves the system performance. When the number or receive antennas is increased from four to eight, the required SNRs for 10% and 1% WER are reduced to 4 dB and 6 dB, respectively. Therefore, MIMO-OFDM is a promising technique for highly spectrally efficient wideband transmission.     

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     

The error performance of CD900-like cellular mobile radio systems

The symbol error performance of CD900-like digital cellular mobile radio systems over narrowband and urban wideband transmission channels was investigated. The basic performance is presented for Gaussian, flat-fading Rayleigh, and log-normal channels in the presence of selection and ratio combining space diversity schemes. For wideband channels having more than one resolvable fading path, a CD900-like system without diversity reception suffers from large residual symbol error probabilities P_R(≈10^-1). The introduction of adaptive correlation diversity (ACD) mitigates the effects of multipath, yielding a P_R of 6×10^-5. Although this P_R value is relatively low, the probability of symbol error (P_e) versus signal-to-noise ratio (SNR) is significantly poorer than for the Gaussian channel. By combining the ACD scheme with space diversity, the P_R is eliminated by P_e >10^-5, and the channel SNR is within 5 dB of the Gaussian channel performance when P_e is 10^-10     

一种新的8~32kb/s宽带嵌入式变速率语音编码方法

 本文基于ACELP和TCX编码技术,提出了一种8~32kb/s五层宽带嵌入式变速率语音编码方法,其中,前三层采用ACELP实现了8kb/s、12kb/s和16 kb/s的嵌入式编码,后两层采用TCX技术实现了24 kb/s和32 kb/s嵌入式编码.实验结果表明,该嵌入式语音编码方法的质量在纯净语音、办公室噪声和层间转换方面接近于ITU-T G.VBR的TOR要求.     

Quality of service and bandwidth efficiency of cellular mobile radio with variable bit-rate speech transmission

The performance of an adjustable source/channel codec in a cellular mobile-radio environment is investigated. The speech transmission rate and the amount of forward error correction change in response to changing channel conditions. The channel rate is constant at 32 kb/s, and when the channel is good all of these bits are used for speech transmission. In intermediate and poor channels the speech rate is 24 or 16 kb/s, and the remaining channel symbols are used for forward error correction. Relative to conventional transmission this approach offers an improved grade of service. For example, the outage rate (the proportion of "poor or worse" communications) goes from nine percent with fixed-rate to three percent with variable-rate transmission. Alternatively, this improved grade of service can be exchanged for higher bandwidth efficiency. The fixed-rate system (with nine percent outage) has 23 users per cell. With 52 users per cell the outage of the variable-rate system is only six percent.     

Quality Analog Scramblers Using Frequency-Response Masking Filter Banks

A very low complexity frequency-domain analog speech scrambler having high-quality speech reconstruction is proposed. The scrambler uses frequency-response masking filter banks which have low computational complexity and narrow transition widths. The proposed system does not require bandwidth expansion, and is robust for transmission channels with distortion characteristics. Preliminary investigations showed that speech scrambled using this method is nearly unintelligible to the casual eavesdropper.     

1.8kb/s的多带线性预测语音编码算法

设计了一种数码率为１．８ｋｂ／ｓ的多带线性预测（ＭＢＬＰ）语音压缩编码算法。该算法采用基于谐振结构的线性预测分析和对激励信号采用多带处理的方法。试验结果表明，本算法提供了相当于码率为２．４ｋｂ／ｓ美国联邦声码器标准ＭＥＬＰ的重建语音质量，具有较高的清晰度和自然度。