Asymptotic receiver structures for joint maximum likelihood timedelay estimation and channel identification using Gaussian signals

The problem of jointly estimating the relative time delay and the impulse response linking two received discrete-time Gaussian signals is addressed. Using two different methods, possible structures for the joint maximum-likelihood (ML) estimator are proposed, when the observation interval is long compared to both the delay to estimate and the correlation time, of the various random processes involved. These structures generalize the cross-correlation method with prefiltering that implements the ML estimation of pure time delays     

Efficient distance measurement method for turbo codes that use structured interleavers

This letter presents an efficient and accurate distance measurement method for tail-biting turbo codes that use structured interleavers. This method takes advantage of the structure in the interleaver as well as the circular property of tail-biting. As such, it significantly reduces the computational complexity, which allows the accurate determination of high minimum distance (d/sub min/) in reasonable time. The efficiency of this method is demonstrated by its ability to determine the true d/sub min/ of 51 and the corresponding true multiplicities for a rate-1/3 turbo code that uses the UMTS 8-state polynomial generators and an MPEG-sized interleaver (1504 information bits) in reasonable time.     

Partial-Response Signaling

This paper presents a unified study of partial-response signaling (PRS) systems and extends previous work on the comparison of PRS schemes. A PRS system model is introduced which enables the investigation of PRS schemes from the viewpoint of spectral properties such as bandwidth, nulls, and continuity of derivatives. Several desirable properties of PRS systems and their relation to system functions are indicated and a number of useful schemes, some of them not previously analyzed, are presented. These systems are then compared using as figures of merit speed tolerance, minimum eye width, and signal-to-noise ratio (SNR) degradation over ideal binary transmission. A new definition of speed tolerance, which takes into account multilevel outputs and the effect of sampling time, is introduced and used in the calculation of speedtolerance figures. It is shown that eye width, a performance measure that has not been used previously in comparing PRS systems, can be calculated analytically in many cases. Exact values as well as bounds on the SNR degradation for the systems under consideration are presented. The effect of precoding on system performance is also analyzed.     

Design, implementation, and characterization of an optical power supply spot-array generator for a four-stage free-space optical backplane

The design and implementation of a robust, scalable, and modular optical power supply spot-array generator for a modulator-based free-space optical backplane demonstrator is presented. Four arrays of 8 x 4 spots with 6.47-mum radii (at 1/e(2) points) pitched at 125 mum in the vertical direction and 250 mum in the horizontal were required to provide the light for the optical interconnect. Tight system tolerances demanded careful optical design, robust optomechanics, and effective alignment techniques. Issues such as spot-array generation, polarization, power efficiency, and power uniformity are discussed. Characterization results are presented.     

Tandem-free VoIP conferencing: a bridge to next-generation networks

This article surveys approaches to teleconferencing in voice over IP networks. The considerations for conferencing include perceived quality, scalability, control, and compatibility. Architectures used for conferencing range from centralized bridges to full mesh. Centralized conference bridges used with compressed speech degrade speech quality when multiple talkers are mixed and subjected to tandem coding operations. Full mesh and multicast solutions (mixing at the end-points) are inappropriate when the number of conferees is large. This article discusses a hybrid solution that incorporates tandem-free bridging (the bridge selects and forwards packets) and endpoint mixing.     

Perceptual coding of narrow-band audio signals at low rates

This paper describes a coding paradigm using coding tools based on the characteristics of the human hearing system so as to accommodate a wide range of narrow-band audio inputs without annoying artifacts at low rates (down to 8 kb/s). The narrow-band perceptual audio coder (NPAC) employs a variety of algorithms to account for the perceptually irrelevant parts of the input signal in addition to statistical redundancies. The new algorithms used in the NPAC coder include a perceptual error measure in training the codebooks and selecting the best codewords which takes into account the audible parts of the quantization noise, a perception-based bit-allocation algorithm and a new predictive scheme to vector quantize the scale factors. The NPAC coder delivers acceptable quality without annoying artifacts for most narrow-band audio signals at around 1 bit/sample. Informal subjective tests have shown that the NPAC coder outperforms a commercial low-rate music coder operating at 8 kb/s.     

A low delay 16 kb/s speech coder

A code tree generated by a stochastically populated innovations tree with a backward adaptive gain and backward adaptive synthesis filters is considered. The synthesis configuration uses a cascade of two all-pole filters: a pitch (long time delay) filter followed by a formant (short time delay) filter. Both filters are updated using backward adaptation. The formant predictor is updated using an adaptive lattice algorithm. The multipath (M, L) search algorithm is used to encode the speech. A frequency-weighted error measure is used to reduce the perceptual loudness of the quantization noise. The addition of the pitch filter gives 2-10-dB increase in segSNR (segmental signal-to-noise ratio) in the voiced segments. Subjective testing has shown that the coder attains a subjective quality equivalent to 7 b/sample log-PCM (pulse code modulation) with an encoding delay of eight samples (1 ms with an 8-kHz sampling rate)     

Shaping multidimensional signal spaces. I. Optimum shaping, shellmapping

The structure of the regions which provide the optimum tradeoff between γ_s (shaping gain) and CER (constellation-expansion ratio) and between γ_s and PAR (peak to average power ratio) in a finite dimensional space is introduced. Analytical expressions are derived for the corresponding tradeoff curves. In general, the initial parts of the curves have a steep slope. This means that an appreciable portion of the maximum shaping gain, corresponding to a spherical region, can be achieved with a small value of CER_s, PAR. The technique of shell mapping is introduced. This is a change of variable which maps the optimum shaping region to a hypercube truncated within a simplex. This mapping is a useful tool in computing the performance, and also in facilitating the addressing of the optimum shaping region. Using the shell mapping, a practical addressing scheme is presented that achieves a point on the optimum tradeoff curves. For dimensionalities around 12, the point achieved is located near the knee of the corresponding tradeoff curve. For larger dimensionalities, a general shaping region with two degrees of freedom is used. This region provides more flexibility in selecting the tradeoff point     

Localization of wideband signals using least-squares and totalleast-squares approaches

In this paper, we introduce a new focusing technique for localization of wideband signals. Relaxing the unitary assumption for the focusing matrices, we formulate the least-square (LS) and the total least-square (TLS) coherent signal-subspace methods. The TLS is an alternative to the conventional LS and uses the fact that errors can exist both in the focusing location matrix as well as in the estimated location matrix at a given frequency bin. To prevent the focusing loss, we use a class of focusing matrices that are constant under multiplication by their Hermitian transpose. The class of unitary matrices comports with this property. We then develop a new focusing technique based on a modification to the TLS (MTLS). It is shown that the computational complexity of the new technique is significantly lower than that for the rotational signal subspace method (RSS). The focusing gain of the new technique is also larger than the focusing gain of the RSS algorithm. The simulation study shows that, compared with the RSS, the new algorithm has a smaller resolution signal to-noise ratio (SNR)