In this study, the performance of two asynchronous multi-access direct sequence code division multiple access (DS-CDMA) and multicarrier code division multiple access (MC-CDMA) systems with binary and polyphase long sequences is compared. An indoor multipath powerline channel model generated from random impedance networks is used. The powerline background, narrowband and impulsive noise are modelled statistically, based on models and measurements done in previous works. A complete analytical performance study is done for asynchronous MC-CDMA employing an interval guard in the form of a cyclic prefix (CP) and periodic long sequences, which provides a better channel coding than short sequences. The analytical results are completed with Monte Carlo simulations for both, DS-CDMA and MC-CDMA systems. Under impulsive noise an upper bound in the bit error rate (BER) performance for DS-CDMA and MC-CDMA systems is obtained analytically, which is confirmed by the Monte Carlo simulations.

One of the recent trends in mobile communication industry is the use of multiple antennas in transmitter and at the receiver in order to gain diversity and achieve high bit rates. At the same time, as the mobile terminals are getting smaller, the number of electronic elements squeezed inside of them increases. Inside this hostile environment the coupling, which can considerably alter the overall system performance, is bound to emerge. To combat the multi antenna RF front-end coupling a DSP software module, positioned between the output of AD converter and any further digital signal processing is proposed. The coupling is modelled with polynomials limited to third order nonlinear behaviour. The inverse nonlinear coupling function of the 9thdegree is solved using simulated annealing (SA) approximation method. The proposed coupling cancellation algorithm is analyzed under different system conditions with different processing power requirements. A trade off between the guaranteed precision and speed is made, and the simplified algorithm version, showing good coupling cancellation characteristics, is presented.

In this paper we address the implementation on FPGAs of a 4G equaliser for a multiple-input multiple-output (MIMO) receiver. It is embedded in a multi-carrier code-division multiple-access (MC-CDMA) radio system, which is able to handle up to 32 users and provide transmission bit-rates up to 125 Mbps. We provide details on decisions taken for the design of the signal processing algorithm, including floating-point to fixed-point translation and architectural considerations. Implementation results using Xilinx Virtex-4 devices are finally reported

Continuous phase modulation schemes, such as Gaussian minimum-shift keying (GMSK), are frequently used with limiter-discriminator (LD) detectors. This paper studies how the side information derived from the signal envelope can enhance the performance of a Viterbi algorithm (VA)-based receiver operating on the LD output of a GMSK scheme. By considering the joint probability density function of envelope and frequency, different approximations yield different novel metrics for VA, using the three-variables envelope and its derivative, and frequency error in different combinations. Simulation results confirm that such envelope-aided VA gives significant performance gains, and that envelope information complements the frequency information output by the LD detector in frequency-selective fading channels.

The conventional decision feedback equalizer (DFE) is a common solution to the frequency selective multipath propagation problem that appears in high data rate radio communications because of its simplicity and good performance. However, when limiter-discriminator (LD) is used for detection of GMSK signals, the global system has a nonlinear characteristic and the conventional DFE yield poor performance even for moderate delay-spread (DS) channels. An envelope-aided DFE is proposed that extends the use of GMSK with LD to high DS channels at a minimum increase in complexity and cost. The performance has been evaluated for both fractionally-spaced (FS) and T-spaced (TS) DFEs, using the exponential power-delay profile channel model.

Maximum likelihood sequence detection (MLSD) using the Viterbi algorithm (VA) is a widely employed detection technique for both coherent and non-coherent detection of continuous phase modulation (CPM) signals due to the inherently nonlinear behaviour of the digital channel. In particular, its application to limiter-discriminator (LD) demodulation schemes has received wide attention in the last years. Somehow, most of the VA proposed, recognize but neglect the effect of the non-Gaussian frequency noise at the output of the LD. As we show in this work this question is closely related to the signal envelope behaviour and hence becomes more important in the presence of frequency selective fading. Two simple schemes that provide a fair solution to that problem are proposed an evaluated in this contribution.

Automatic score transcription goal is to achieve an score-like (notes pitches through time) representation from musical signals. Reliable pitch extraction methods for monophonic signals exist, but polyphonic signals are much more difficult, often ambiguous, to analyze. We propose a computationally efficient technique for automatic recognition of notes from a polyphonic signal. It looks for correctly shaped (magnitude and phase wise) peaks in a, time and frequency oversampled, multiscale decomposition of the signal. Peaks (partial candidates) get accepted/discarded by their match to the window spectrum shape and continuity-across-scale constraints. The final partial list builds a resharpened and equalized spectrum. Note candidates are found by searching for harmonic patterns. Perceptual and source based rejection criteria help discard false notes, frame-by-frame. Slightly non-causal postprocessing uses continuity (across a <150 ms observation time) to kill too short notes, fill in the gaps, and correct (sub)octave jumps.

An algorithm capable of reliably estimating the fundamental frequency of sound signals is described. Pitch estimation is carried out by means of a new loose harmonic matching algorithm which searches the spectrum of a sound for evenly-spaced peak patterns. This unaided algorithm is by itself capable of correctly estimating the fundamental frequency of practically any musical sound. There is also a post-processor-based on fuzzy neural networks-which attempts to map the fundamental frequency (a physical property of the waveform) into pitch (a perceptual variable). The information is transformed into musical data and sent via standard MIDI (musical instruments digital interface) to any other MIDI-controlled device. It has been implemented in real time on a single AT&T's DSP16C processor with practically no additional hardware.>

In this contribution an ASIC capable of performing several digital signal processing functions at high speed is described. Although conceptually simple, the circuit can function as adaptive equalizer, RAKE receiver, correlator and so on. Parallelism, tap selection and some other features contribute to make the processor very efficient. It is capable of carrying out 450 multiply/add operations per second, at a sampling rate of 10 Msamples/sec.>

The authors describe the main topics related to a real-time implementation of the Proposed Federal Standard 1016 (PFS-1016) 4800 bps CELP (code excited linear prediction) voice coder together with some alternatives to extend real-time CELP schemes over a wide range of bit rates (8000, 6500, and 2400 bps are considered). Several procedures are evaluated in order to select the options in PFS-1016 that provide the highest quality for an implementation based on a single AT&T DSP32C chip. From the basic CELP scheme, higher and lower bit rates are derived after evaluation of the efficiency obtained for the different elements in this speech coding algorithm.>

The main results from work on analysis techniques and quantization procedures for real-time implementation of the Proposed Federal Standard 1016 4800-b/s voice coder developed by the US DoD and AT&T Bell Laboratories are presented. Reduced search procedures are described for the adaptive and stochastic codebooks, two basic functions in code-excited linear predictive coders (CELPs), where great computational effort is required. An efficient procedure for the joint computation and quantization of the line spectrum pair (LSP) is also proposed. Details about a preliminary real-time implementation on a commercial board with a single DSP32C processor are described.>

A speech coder is presented which combines vector excitation coding (VXC) with frequency-domain representations so as to obtain a high-quality efficient scheme at 4.8 kb/s based on the use of different coding strategies and bit allocations for different frames of speech and different frequency bands. Attention is focused on the representation of voiced sounds and on multiband procedures for representing the short-time Fourier transform. The proposed scheme starts with a phonetically based frame segmentation. Multiband long-term prediction is included only when necessary, and random excitation in voiced sounds is spectrally shaped by using a linear predictive coding envelope from the long-term prediction error.>

The authors discuss various possible synthesis procedures for frequency-domain vector-excited coding (VXC). The algorithm, called vector adaptive transform coding (VATC), is a two-stage vector quantizer for the short-time Fourier transform (STFT) of speech. One stage represented by an adaptive codebook based on long-term prediction and the other based by a random codebook. The authors focus on the differences between time and frequency VXC, stressing the importance of the long-term prediction. A fast algorithm for long-term prediction is introduced, and different strategies for noise reduction are proposed. VATC is presented as a particular case of a general vector linear prediction (VLP) scheme. Following this approach, a coder based on a self-spectral shaping procedure is proposed. This coder has the potential of producing telephonic-quality speech at rates lower than 4.8 kb/s.>

A frequency-domain search procedure is presented that requires a computational load of approximately 1.5 MFLOPS, reduces the storage cost, and maintains the speech quality. A vector-excitation-coding (VXC) method in the transform domain called vector adaptive transform coding (VATC) is proposed. VATC provides a general framework for VXC in the transform domain, and the quality for the synthetic speech is close to that obtained in code-excited line prediction (CELP). The major advantages of VATC are the different ways of including information from previous frames, different perceptual distortion measurements, different transformations, efficient search algorithms, and the use of vector predictive quantization techniques so as to have a fully quantized coder at 4.8 kb/s.>

Both multipulse and stochastic coders render telephonic, or close to telephonic quality, for speech signals. Stochastic coders are more efficient from the point of view of bit rate and multipulse coders are the most efficient ones if we are looking for low computational cost. In this paper we study some possibilities of coding speech using the multipulse model but as efficiently as the stochastic coders does. Synthetic speech is generated from a multipulse sequence, obtained from a codebook, through a LPC syntesis filter. Then the model for speech signal is the multipulse model; it resembles the stochastic one in that the input sequence is obtained from a codebook but differs since it does not use pitch filter.

The bit rates of existing speech coders can be scaled down through the use of time-compression of speech signals. Of course this implies a small increase in coder complexity and some loss of quality. This work focuses on the use of dynamic time-compression in order to reduce the loss of quality. Long, stationary segments do exist in speech, therefore they can tolerate high compression rates without much degradation. Use of the appropriate rate everywhere in time results in increased overall performance from the point of view of both total time-compression factor and resulting signal-to-error ratio.

In this paper, we propose a simplification of the model introduced by B.S. Atal and M.R. Schroeder for a code-excited linear predictor. Our scheme is based on a two-step search of the optimum code-word. In the first step, a sub-set in the code-book is selected by using a simple distance measure and taking into account the selections already made for the previous frames of speech. In the second step, the original model of Atal and Schroeder, including perceptual weighting, is applied to the sub-set in order to make the final selection of the optimum code-word. The proposed scheme reduces the computational load involved in the original one, and our preliminar results indicate that the quality of the synthetic speech is fairly good.

This paper presents a new adaptive algorithm for the long term stable operation of a digitally implemented fractionally-spaced adaptive equalizer. We show how previously reported solutions may become useless working with a too small word size. Then, we propose an alternative based on performing an equalizer tap weight normalization after each new adaption. Thus, we simultaneously prevent register overflows an avoid the SNR degradation when applying the previous algorithms.