"Bit Loading with BER-Constraint for Multicarrier Systems",
IEEE Trans. Wireless Commun., vol. 4, no. 4, pp. 1383-1387, July 2005.A. M. Wyglinski, F. Labeau, and P. Kabal
"Bit Loading with BER-Constraint for Multicarrier Systems",
IEEE Trans. Wireless Commun., vol. 4, no. 4, pp. 1383-1387, July 2005.
We present discrete adaptive bit loading algorithms for multicarrier systems with uniform (nonadaptive) power allocation operating in a frequency selective fading environment. The algorithms try to maximize the overall throughput of the system while guaranteeing that the mean bit error rate (BER) remains below a prescribed threshold. We also study the impact of imperfect subcarrier signal-to-noise ratio information on throughput performance. Results show that the proposed algorithms have approximately the same throughput and mean BER as the optimal allocation while having a significantly lower computational complexity relative to other algorithms with near-optimal allocations. Moreover, when compared with algorithms that employ approximations to water filling, the computational complexity is comparable while the overall throughput is closer to the optimum.
A. M. Wyglinski, P. Kabal, and F. Labeau
"BER-Constrained Loading Algorithms for
Multicarrier Spatial Diversity Systems",
Proc. IEEE Vehicular Technol. Conf. - Fall (Dallas, TX), 5 pp. Sept. 2005.
In this paper, four novel loading algorithms are proposed for multicarrier systems employing spatial diversity. The objective of these algorithms is to increase system throughput while ensuring the mean bit error rate (BER) is below a specified limit. A subcarrier signal is transmitted simultaneously on a subset of antennas, passed through a multiple-input multiple output (MIMO) channel, intercepted by a subset of receive antennas, and linearly combined. Both antenna subset configurations may differ for each subcarrier. The rationale for this form of antenna selection is to efficiently distribute power across the antennas and to reduce array processing complexity. Bit loading is employed by two of the proposed algorithms to further increase throughput. Simulation results show a tradeoff between the allocation algorithm computational complexity and system throughput.
A. M. Wyglinski, M. Cudnoch, F. Labeau, and P. Kabal
"Practical Termination Strategies for
Subcarrier Equalizer Tap Loading Algorithms",
Proc. IEEE Vehicular Technol. Conf. - Fall (Dallas, TX), 5 pp. Sept. 2005.
In this paper, a termination strategy for subcarrier equalizer tap loading algorithms is proposed. The objective of these algorithms is to non-uniformly distribute equalizer taps incrementally across the subcarriers of a multicarrier system to reduce the overall distortion introduced by the channel. Knowing when enough equalizer taps have been allocated is the task of the algorithm’s termination strategy, which employs a number of criteria in the decision process. The proposed termination strategy, which is based on limiting the total number of taps used by the system, is presented and compared with three other strategies proposed earlier. Simulation results for a system employing the proposed strategy using different total tap limits emphasize the advantages of equalizer tap loading algorithms over conventional multicarrier equalization schemes.
"Classified Highband Excitation for
Bandwidth Extension of Telephony Signals",
Proc. European Signal Processing Conf. (Antalya, Turkey), 4 pp., Sept. 2005.
Current telephone networks compromise bandwidth for efficiency. The impairment of the audio quality in telephony has become a problem for the rapidly emerging sophisticated wideband telecommunications systems. We present a classified bandwidth extension algorithm which recovers the missing highband portion of telephony signals. We describe a new highband excitation generator, a Pitch-Synchronized-BandPass-Shifted-Sum excitation for strongly harmonic signals such as some voiced phonemes or some music audio signals. For other signals, a BandPass Envelope Modulated Gaussian Noise is used as the highband excitation. The highband spectrum envelope and the excitation gain are estimated using classified Gaussian Mixture Models. Objective measurements of spectrum sections and informal subjective tests of both reconstructed telephony speech and audio signals show more highband harmonic textures for strongly-harmonic signals than previous bandwidth extension methods.
T. Leppert, F. Labeau, and P. Kabal
"Performance of Noise-Shaping in
Oversampled Filter Banks", Proc. European Signal Processing Conf.
(Antalya, Turkey), 4 pp., Sept. 2005.
The use of a noise shaping system in oversampled filter banks has been shown to improve the effective resolution of subband coders. In this paper, a brief review of past work is given and a study of the effects of the characteristics of the employed filter banks is presented. It is shown that an increase in filter length and an increase in the degree of overlap between neighboring channels contribute independently to a better performance. Also, it is shown that near-perfect reconstruction filter banks are limited by their reconstruction error but yield good results at a low bitrate. This is supported by both theoretical and experimental results.
"Dynamically Adding Redundancy for
Improved Error Concealment in Packet Voice Coding", Proc. European Signal
Processing Conf. (Antalya, Turkey), 4 pp., Sept. 2005.
This paper presents a method to improve the performance of redundancy-based packet-loss-concealment (PLC) schemes. Many redundancy-based PLC schemes send a fixed amount of extra information about the current packet as part of the subsequent packet, but not every packet is equally important for PLC. We have developed a method to determine the importance of packets and we propose that redundant information should only be sent for the important packets. This results in a lower average bit-rate compared to sending a fixed amount of extra information, without sacrificing much from the quality of the concealment. We use a linear prediction (LP) based speech coder (ITU-T G.723.1) as a test platform and we propose that only the excitation parameters should be sent as extra information since LP parameters of a frame can be estimated using the LP parameters of the previous frame.
W. Chen, P. Kabal, and T. Z. Shabestary
"Perceptual Postfilter Estimation for Low Bit Rate Speech
Coders Using Gaussian Mixture Models", Proc. Interspeech 2005
(Lisbon, Portugal), pp. 3161-3164, Sept. 2005.
A novel perceptual postfilter is introduced. For each frame, the filter gains, z, are estimated given a vector, y, of the quantized LSFs and the long-term prediction gain of the corresponding frame. The proposed perceptual postfilter is derived from an optimal MMSE estimator, i.e. the estimated gain vector is z' = E{z|y}. The MMSE estimator is based on the conditional pdf of z given y, which is computed from the joint pdf modelled by a GMM. The proposed perceptual postfilter improves the speech naturalness comparing with the conventional adaptive postfilter, while maintaining the property of being an "add-on" postfilter without modification to the current encoder.
T. H. Falk, W.-Y. Chan, and P. Kabal
"An Improved GMM-Based Voice Quality Predictor",
Proc. Interspeech 2005 (Lisbon, Portugal), pp. 2733-2736, Sept. 2005.
A voice quality prediction method based on Gaussian mixture models (GMMs) is improved by constructing a feature selection algorithm to provide the best GMM-based prediction quality. The proposed sequential selection algorithm performs N-survivor search, allowing for trading between design complexity and performance. Simulation shows that predictors designed using the proposed algorithm outperform two benchmark selection algorithms. Performance improvements over the ITU-T P.862 PESQ standard are also attained.
A. M. Wyglinski, F. Labeau, and P. Kabal
"Loading Algorithms for Multicarrier Spatial
Diversity Systems employing Antenna Subset Selection", Proc. IEEE Pacific
Rim Conf. Commun., Computers, and Signal Processing (Victoria, BC), pp.
490-493, Aug. 2005.
This paper presents two novel loading algorithms for multicarrier systems performing spatial diversity. The algorithms use bit allocation techniques to increase throughput while ensuring the system remains below a specified error rate. One algorithm uses the same signal constellation across all subcarriers while the other varies the signal constellations. To reduce hardware costs, power consumption, and complexity, the proposed algorithms simultaneously perform antenna subset selection, choosing array configurations that yield further increases in the overall throughput. Our results show that the combination of multiple antennas and bit allocation by the proposed algorithms substantially increases throughput. Furthermore, hardware costs are reduced with a negligible penalty in throughput when antenna subset selection is employed.
Y. Ould-Cheikh-Mouhamedou, S. Crozier, and P. Kabal
"Comparison of Distance
Measurement Methods for Turbo Codes", Proc. Canadian Workshop
Inform. Theory (Montreal, QC), pp. 36-39, June 2005.
Reliable distance measurement methods for turbo codes are a key element in the design of interleavers with high minimum distances, which are essential for lowering the flare effect at low error rates. The usefulness of such methods depends strongly on their computational complexity, especially for long interleavers with high minimum distances. This paper improves the reliability of the double-impulse iterative decoding method and compares it with Garello’s true minimum distance method, the error-impulse method and the all-zero iterative decoding method. The comparison is based on the interleavers specified in the Digital Video Broadcasting with Return Channel via Satellite (DVB-RCS) standard, random interleavers and dithered relative prime (DRP) interleavers. A new interleaver for an MPEG packet of size 1504 information bits is designed for the DVB-RCS standard. The new interleaver provides an improvement of at least 0.4 dB at low error rates.
R. Der, P. Kabal, and W.-Y. Chan
"Rate-Distortion Allocation for Time-Frequency
Dependent Audio Coding", Proc. IEEE Int. Conf. Acoustics, Speech, Signal Processing
(Philadelphia, PA), pp. III-197-III-200, March 2005.
A stream coding framework is presented for solving the distortion-constrained time-frequency dependent quantization problem that naturally arises when overlapped time-frequency decompositions are used. The main contributions of this paper are (1) an efficient rate-distortion allocation algorithm for dependent quantization when the neighborhood of dependency is large; and (2) demonstration that a perceptual Excitation Distortion measure produces better coded audio quality than the conventional Noise-to-Mask Ratio measure.
Paper titles.