The performance in terms of signal-to-interference ratio (SIR), teletraffic, and spectral efficiency of a combined macrocellular and microcellular network is investigated when either both types of cells share the same channel set, or when the channel set is partitioned between the macrocells and the microcells. The analysis is for time-division multiple access (TDMA) with frequency hopping, power control, and discontinuous transmission, and the radio channel is composed of an inverse fourth-power path loss law with log-normal fading. We commence by introducing a single microcell into a hexagonal cluster of macrocells before considering clustered microcells. Both omnidirectional and sectorized cells are examined. We find that high reuse factors are required when channel sharing is employed. When channel partitioning is used, no co-channel interference occurs between the microcells and the macrocells allowing them to be planned independently. The reuse factors in the microcells and macrocells therefore do not need to be increased beyond conventional values. The outcome is that by opting for channel partitioning, the improvement in spectral efficiency compared to channel sharing is two to three times greater.

Quadrature amplitude modulation (QAM) schemes which vary the number of modulation levels in accordance with the mobile radio fading channel variations are investigated. Important parameters considered are the fading rate and the block size used. We describe how the adaptive QAM modems can be employed and consider their use in a DECT-like TDD packet structure. System performance in the presence of cochannel interference is also considered. Simulations show that the variable rate system has about 5 dB improvement in channel SNR over a fixed 16-level QAM system for BER's between 10/sup -2/ and 10/sup -5/ and channel SNR's between 25 and 40 dB.>

The audio quality, robustness and complexity issues of a novel mobile digital audio broadcast (DAB) scheme are addressed. The audio codec is based on a combination of subband coding (SBC) and multipulse excited linear predictive coding (MPLPC), where the bit allocation is dynamically adapted according to both the signal power in different subbands and a perceptual hearing model. Typically a segmental signal to noise ratio (SEGSNR) in excess of 30 dB associated with high fidelity (hi-fi) subjective quality was achieved for 2.67 bits/sample transmissions at a mono bit rate of 86 kbits/s. Four different source-matched forward error correction (FEC) schemes were investigated in order to explore the complexity, bit rate and robustness trade-offs. When using 4 bit/symbol 16-level star-constellation quadrature amplitude modulation (16-StQAM) the overall signalling rate became approximately 30 kBaud, accommodating two stereo DAB channels in a conventional 200 kHz analogue FM channel's bandwidth. The diversity assisted DAB scheme required a channel signal to noise ratio (SNR) of about 25 dB for unimpaired audio quality via the worst-case Rayleigh fading mobile channel, when the mobile speed was 30 mph and the propagation frequency was 1.5 GHz. In case of the stationary Gaussian scenario an SNR of about 20 dB was required.>

The complexity robustness, image and speech quality as well as packet multiplexing issues of a re-configurable multi-media mobile communicator are addressed. The proposed moderate complexity motion-compensated discrete cosine transform (DCT) based image communicator provides an image peak signal-to-noise ratio (PSNR) of 38 dB at an average bit rate of about 25 kbits/s. The speech codec used is a low-complexity 32 kbit/s CCITT G721 standard scheme. Bandwidth efficient 16 or 64-level quadrature amplitude modulation (QAM) combined with embedded low-complexity binary Bose-Chaudhuri-Hocquenghem (BCH) forward error correction (FEC) coding is deployed. The 2O-slot packet reservation multiple access (PRMA) scheme used supports an extra 2.4 kbit/s low-rate data channel for each speech user, in addition to providing 5-6 videophone channels. The ADPCM/DCT/BCH/16-QAM and ADPCM/DCT/BCH/64-QAM schemes provide nearly unimpaired speech and image quality for channel SNRs in excess of 30 dB and 38 dB, respectively.>

A combination of a decision feedback equalizer (DFE) and a Viterbi processor for the equalization of continuous phase modulated (CPM) signals transmitted over wideband mobile radio channels is described. The equalizer structure allows soft-decision to be made and incorrect decisions to be changed in the feedback filter of the DFE. The provision of soft decisions significantly alleviates the catastrophic effect of error propagation due to error bursts caused by the fading channel. Simulation results show that the proposed equalizer approaches the performance of the more complicated maximum likelihood sequence estimation (MLSE) detector to within 2.0 dB and 2.5 dB at bit error rates (BERs) of 10/sup -2/ and 10/sup -3/ in a two-path Rayleigh fading channel while the complexity is reduced to half of that of the MLSE detector.>

The feasibility of a mobile video handset is investigated for Rayleigh-fading channels, where transmissions must be confined to the channel's coherence bandwidth to avoid the deployment of complex high-power-consumption channel equalizers. This necessitates the utilization of a low-bit-rate image codec error-protected by embedded low-complexity BCH codecs and spectrally efficient 16-level quadrature amplitude modulation (16 QAM). Motion-compensated nonuniform seven-band subband coding with subband-specific scanning, adaptive quantization, runlength coding, and adaptive buffering to equalize bit-rate fluctuations offer good objective and subjective image quality at moderate complexity and a bit-rate of 55 kb/s. Using a twin-class embedded-BCH error protection as well as pilot symbol 16 QAM and diversity assisted, the 22 kBd candidate system yields unimpaired image quality for average channel signal-to-noise ratios (SNRs) in excess of about 16-18 dB, when the mobile speed is 4 miles/h.>

A modified regular-pulse excitation (RPE) linear predictive coding (LPC) codec is used to encode audio signals of 15 kHz bandwidth. With appropriate perceptual masking and carefully selected excitation structure, a fixed-pulse excitation (FPE) LPC codec is able to provide high-fidelity audio output. The quantization of both the LPC parameters and the excitation pulse sequence is examined. It is found that the LPC parameters can be quantized efficiently by adaptive differential PCM, while the excitation pulse sequences need more bits to quantize the amplitudes compared to narrowband speech encoding. Informal listening tests show that high-fidelity audio can be achieved at bit rates less than 100 kb/s.>

The performance of error protected transformed binary pulse excited (TBPE) coders over Rayleigh fading channels is investigated. The TBPE coder is a stochastically excited linear predictive coding (LPC) coder which produces near-toll quality speech in the vicinity of 8 kb/s utilizing an extremely simple excitation search procedure. A TBPE coder operating at 7.4 kb/s is used, and two methods of embedding channel coding into the speech coder are studied, aiming at the overall bit rate of 11.4 kb/s specified for the half-rate GSM coder. A method of embedding Reed-Solomon codes into speech coding is presented, and it is compared with rate-compatible punctured convolutional (RCPC) codes. Results show the superiority of the proposed scheme over RCPC codes in both performance and implementation simplicity.>

The teletraffic performance of a highway microcellular digital mobile radio system having an oversailing macrocell that spans many microcells is presented. The microcellular cluster is composed of concatenated segments of the highway where each segment is a microcell, typically 500-2000 m in length, with the base stations located at lamp-post elevations. A narrowband time-division-multiple-access arrangement supporting ten channels per carrier and one carrier per base station is used. The teletraffic analysis assumes there are n-up and n-down lanes, and that the vehicular speeds conform to a truncated Gaussian distribution whose mean speed is 100 or 50 km/h when the vehicular traffic is free-flowing or in traffic-congested conditions, respectively.>

Three systems are proposed for embedding data into industrial quality monochrome analog pictures. The video signal on each scan line is sampled, and a data bit is inserted into a block of three or five pels by modulo masking scrambling the luminance level of only one pel in the block. Prior to transmission, the combined data and video sequence is converted into a continuous signal with a bandwidth that is no greater than that of the original video signal. Using six images each containing 65 536 pels, Systems 1 and 2 embedded an average of 17 430 and 8713 bits per image, while System 3 accommodated data at a constant rate of 21 760 bits/image. The data embedding procedures of Systems 1, 2, and 3 operated with average picture SNR's of 41, 44, and 30 dB, respectively, when the transmission channel was ideal. When the transmission was over a channel composed of a second-order Butterworth filter plus additive noise that yield a channel SNR of 40 dB, no bit errors occurred but System 3 offered the greater safety margin to bit errors than Systems 1 and 2.

Telephone channels restrict the bandwidth of speech signals to approximately 0.3 to 3.3 kHz, with the consequence that the intelligibility of unvoiced sounds may be significantly impaired. To prevent this bandlimitation of unvoiced sounds while still confining the speech to the telephonic bandwidth, we propose a scheme which on recognizing the presence of unvoiced sounds extending to 7.6 kHz, frequency maps them into the band 0.3 to 3.3 kHz. Four mapping laws are considered and the unvoiced speech is compressed using each law. Frequency de-mapping is employed, and the law that has the best spectral match to the speech spectrum is selected. Voiced speech is bandlimited from 0.3 to 3.3 kHz. Results indicate that the adaptive frequency mapping algorithm significantly enhances the recovered speech compared to telephonic speech.

Telephone channels restrict the bandwidth of speech signals to approximately 0.3-3.3 kHz, with the consequence that the intelligibility of unvoiced sounds may be significantly impaired. To prevent this band limitation of unvoiced sounds while still confining the speech to the telephonic bandwidth, we propose a scheme which, on recognizing the presence of unvoiced sounds extending to 7.6 kHz, frequency maps them into the band 0.3-3.3 kHz. Four mapping laws are considered and the unvoiced speech is compressed using each law. Frequency demapping is employed, and the law that has the best spectral match to the speech spectrum is selected. Voiced speech is band limited from 0.3 to 3.3 kHz. Results measured over 16 ms, a phoneme, and word durations indicate that the adaptive frequency mapping algorithm significantly enhances the recovered speech compared to telephonic speech. Informal listening experiences support these findings.

A new diversity technique is proposed to combat Rayleigh fading in digital mobile radio systems transmitting speech signals. The speech signals are μ-law PCM encoded (\mu = 255, 8 kHz sampling, 8 bits/code word, 64 kbit/s data rate), and alternate data words are used to form two streams called "odd" and "even." The even stream is delayed by τ seconds and the streams are interleaved prior to radio transmission using two-level PSK modulation. At the receiver the odd data stream is delayed by τ and interleaved with the even stream. Consequently, if an error burst occurs, the effect of the reshuffling of the data stream is, in general, to place words with bit errors in juxtaposition to those correctly received. After μ-law PCM decoding of the words, a statistical error detection strategy is evoked to identify the erroneous samples. These samples are replaced by adjacent sample interpolation to give the recovered speech sequence. No recourse to channel protection coding is made. In our experiments a Rayleigh fading envelope was generated from a hardware simulator and stored in a computer, along with four sentences of speech. The system was then simulated and the recovered speech perceived. The objective performance measures were segmental SNR for the audio signal, and BER. Different error detection strategies were examined and restrictions on τ investigated. For a mobile speed of 30 mph, SNR values of 32, 21, and 16 dB were obtained for BER values of 0.1, 1, and 2 percent, corresponding to SNR gains over an uncorrected system of 3, 9, and 11 dB, respectively.

A method of embedding data into speech signals is proposed. The speech signal is scrambled using the data as the scrambling key, while the receiver adopts the role of a code breaker. By judicious choice of scrambling algorithm the receiver can be made to break the code at every attempt. We found that 126 b/s can be transmitted without error over a channel whose additive noise is only 10 dB below the mean square value of the speech signal.

The sequential gradient estimation predictor is compared in detail to the stochastic approximation predictor, and both are evaluated in an ADPCM codec. A switched predictor having two coefficients is then described for use in a DPCM-AQF codec. This predictor divides the range of the correlation coefficient of the speech signal into zones, and as the correlation coefficient changes zones, the predictor coefficients undergo a substantial modification. By this method the adaptation rate of the predictor is improved, particularly during transitions between unvoiced and voiced sounds.

Telephone channels restrict the bandwidth of speech signals to within the range of 0.3 to 3.4 kHz and in doing so cause significant attenuation and distortion in unvoiced sounds with resulting loss of intelligibility and quality. A bandwidth compression system is proposed in this paper, whereby 0.3 to 7.6 kHz band limited speech is frequency mapped into the 0.3 to 3.4 kHz telephonic bandwidth range and is subsequently digitized prior to transmission. The received signal is decoded and spectrally expanded using a corresponding frequency de-mapping process. Consideration is given to the digital encoding of the compressed signals for bit rates between 16 and 32 kbits/s. We find that the quality of the recovered speech is improved compared to band limited telephonic speech.

This paper explores the use of slope limiting filters to mitigate the errors induced in adaptive and non-adaptive PCM and DPCM speech. The two correctors discussed use slope limiting to remove the spikes associated with bit errors. They are treated as add-on systems with no knowledge of the input signal other than that it is a speech waveform, and use very simple replacement algorithms when they detect an anomalous condition. The methods do not make an "optimum" replacement of anomalous conditions; however, simple replacement algorithms result in significant improvement of corrupted speech, especially where listening fatigue is a factor. They use inexpensive (analog and delta modulation) circuits which are easily constructed and installed.

Easy to use simple equations of optimum slope loading factor and peak SNR are given for a linear delta modulator having Gaussian input signals with band-limitedRCspectra. The equations are accurate to within 1 dB over a wide range of normalized sampling frequency and forRCspectrum bandwidths of practical interest.

The statistical block protection coding scheme [2] for protecting DPCM encoded speech signals through noisy channels has been extended to accommodate DPCM-AQF encoded speech, where AQF stands for adaptive quantization with forward (explicit) transmission of step size. Signal-to-noise ratio (SNR) gains, typically 12 dB, have been achieved over a dynamic range> 20dB for a bit error rate (BER) of 1.4 percent and the SNR improvement is found to increase with BER. Perceptual improvements in decoded speech have a good correspondence with the gains in SNR. The penalty for this substantial enhancement in the performance of the DPCM-AQF system is an increase in transmission bit rate of 3.5 percent and encoding delay of 64 ms.

The envelope dynamic ratio quantizer (envelope DRQ) is an adaptive quantizer for speech signals. By utilizing the envelope of the speech, nonlinear elements, a fixed quantizer and a simple predictor, a closed-loop adaptive quantizer emerges having a high constant SNR over a wide dynamic range. The theory of the quantizer is presented, together with computer simulation results which show an improvement.compared to the one word memory APCM system. Finally, the simplicity of implementing the envelope DRQ is described.

A sequential gradient estimation predictor [SGEP] for speech signals is presented. In a given sampling interval, each prediction coefficient in turn is increased and decreased in value by a prescribed amount, while the other coefficients are kept constant. Two predictions are then made and the better enables the coefficient to be modified in the correct direction, but by an amount determined by a number of factors. The superiority of SGEP over the stochastic approximation predictor is illustrated by waveforms and snr (signal to noise ratio) performance curves.

It is possible to detect the presence of transmission errors in the decoded sample sequence of a differential pcm system. The detection procedure is based on the sample-to-sample differences in the received signal. It infers an error whenever an individual difference is substantially greater than the rms difference. On the detection of an error, the sample affected is replaced by the result of a smoothing procedure and subsequent samples are adjusted to compensate for the error propagation inherent in differential pcm. This difference detection and correction (DDC) scheme substantially improves system performance at high error rates.

In pulse code modulation (PCM) of speech, transmission errors cause changes in the statistical properties of the received samples. Recently published work describes computer simulations of a difference detection and correction (DDC) system which detect errors on the basis of these changes. The system examines the differences between adjacent received samples. If a particular difference exceeds a threshold, which is dependent on the rms value of a block of 64 differences, the quantized sample responsible for this large difference is deemed to be erroneous. A nonlinear filter of the median type is then introduced as a corrector. We now present a comprehensive description of the hardware realization of this system, and present measured results of signal-to-noise ratio (snr) as a function of bit error rate, input power and the error criterion for narrow-band white noise input signals. An improvement of 6 dB in snr is achieved for error rates between 0.2 and 2.0%. The snr results are consistent with subjective impressions of the quality of speech processed by the DDC system.

A "statistical average step size," γav, which is a function of the rms power of a band-limited white Gaussian input signal, is defined for high information and first order constant factor delta modulation systems. The signal-to-noise ratios (SNR) are then calculated for these instantaneously adaptive delta modulation systems by substituting γavfor the fixed step size in the formulas for quantization (i.e., granular) and slope-overload noise applicable to linear (single integration) delta modulation systems. The calculated SNRs are in close agreement to those obtained by computer simulations. As γavis difficult to calculate exactly, an empirical expression is found which is applicable to different encoding conditions. This empirically formulated γavcan be used to estimate SNR to an acceptable accuracy for most practical situations. A noise component not present in linear delta modulation is identified and called "overshoot noise" and the cross-correlation ρxebetween input and error signals is studied. The SNR and ρxefor linear delta modulation are presented as reference bases.