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Wilf P. LeBlanc

dblp:142/7692 · DBLP profile ↗
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4ranked-venue papers
2as first author
0since 2021 · last 1994
—ORCID · none

Domains — the database's venue-derived domains; a paper can count in several

Artificial intelligence and machine learning · 1 · 1 first-authorGraphics, computer vision, multimedia, augmented reality and games · 1

Expertise — from the expertise taxonomy: the topics of the expert's papers under the CCF categories. A weight counts papers with recency: 1 for a paper about the topic, 0.3 when the topic is its context, halved every five years.

Artificial intelligence
1 paper
Speech recognition and synthesis · 67% Representation and self-supervised learning · 33%

Topics — the 3 heaviest of 3, each with the papers that count most for it

TopicWeightPapersLastEvidence papers
Natural language and speech › Speech recognition and synthesis › speech coding
linear predictive coding
0.011993
Efficient search and design procedures for robust multi-stage VQ of LPC parameters for 4 kb/s speech coding · IEEE Trans. Speech Audio Process. 1993
Natural language and speech › Speech recognition and synthesis
speech coding
0.011993
Efficient search and design procedures for robust multi-stage VQ of LPC parameters for 4 kb/s speech coding · IEEE Trans. Speech Audio Process. 1993
Machine learning › Representation and self-supervised learning
vector quantization
0.011993
Efficient search and design procedures for robust multi-stage VQ of LPC parameters for 4 kb/s speech coding · IEEE Trans. Speech Audio Process. 1993

Methods — techniques the papers use, named apart from their topics

tree-searched multistage vector quantization · 0.0m-l search · 0.0joint codebook design · 0.0
YearPublicationVenuePosition
1994 Low complexity, low delay speech coding for indoor wireless communications
abstract
The paper presents a low complexity, low delay variable rate speech coding method suitable for low power (CDMA or TDMA) indoor wireless communications. The indoor wireless communications channel is characterized by relatively deep fades resulting in bursty errors. A robust speech codec is required to combat this bursty nature. Low power operation is essential, which negates the applicability of LD-CELP (G.728) for indoor wireless applications, Historically, ADPCM (G.721) has been proposed, which lacks robustness to random and bursty errors and has a relatively high bit rate (32 kb/s). To circumvent the need for relatively complex echo-cancellation, low end to end delays are required which implies that interleaving is not applicable. In the paper, a low complexity, low delay, variable rate CELP algorithm is proposed and techniques to improve the robustness to bursty errors are investigated. The new technique is compared to ADPCM on the basis of complexity and performance (MOS) over measured indoor wireless channels at 1.7 GHz with various combinations of space, frequency and code diversity.>
Wilf P. LeBlanc, Samy A. Mahmoud
VTC1
1994 A low power, single chip realization of a low-speed, low-delay CELP coder/decoder for indoor wireless systems
abstract
32 kb/s adaptive differential pulse coded modulation (ADPCM) is a widely accepted high quality speech coding algorithm. Single chip devices exist for 32 kb/s ADPCM. This algorithm requires a relatively large bandwidth and does not perform exceptionally well in noisy wireless environments. The 16 kb/s low-delay code excited linear prediction (LD-CELP) coding algorithm does perform well in noisy environments and requires a reduced bandwidth. Through algorithm optimization and simulated verification of mixed analog and digital VLSI partitioning a single chip implementing the 16 kb/s LD-CELP voice coding algorithm will be possible. The performance of this coder chip will be comparable to the quality of 32 kb/s ADPCM. In this paper we propose the use of a link between Comdisco's SPW and Synopsis to determine the hardware related performance degradation and the technology/complexity related power dissipation and area parameters. It will both prove the concept of design and determine the VLSI implementation feasibility.>
D. P. Noel, Tad A. Kwasniewski, Samy A. Mahmoud, Wilf P. LeBlanc
VTC4
1993 Efficient search and design procedures for robust multi-stage VQ of LPC parameters for 4 kb/s speech coding
abstract
A tree-searched multistage vector quantization (VQ) scheme for linear prediction coding (LPC) parameters which achieves spectral distortion lower than 1 dB with low complexity and good robustness using rates as low as 22 b/frame is presented. The M-L search is used, and it is shown that it achieves performance close to that of the optimal search for a relatively small M. A joint codebook design strategy for multistage VQ which improves convergence speed and the VQ performance measures is presented. The best performance/complexity tradeoffs are obtained with relatively small size codebooks cascaded in a 3-6 stage configuration. It is shown experimentally that as the number of stages is increased above the optimal performance/complexity tradeoff, the quantizer robustness and outlier performance can be improved at the expense of a slight increase in rate. Results for log area ratio (LAR) and line spectral pairs (LSPs) parameters are presented. A training technique that reduces outliers at the expense of a slight average performance degradation is introduced. The method significantly outperforms the split codebook approach.>
Wilf P. LeBlanc, B. Bhattacharya, Samy A. Mahmoud, Vladimir Cuperman
IEEE Trans. Speech Audio Process.1
1992 Tree searched multi-stage vector quantization of LPC parameters for 4 kb/s speech coding
abstract
The authors present a tree searched multi-stage vector quantization (TS-MSVQ) scheme which achieves spectral distortion lower than 1 dB with low complexity and good robustness using 24 b/frame. The M-L search is used and it is shown that it achieves performance close to that of the optimal search for a relatively small M. The best performance/complexity trade-offs are obtained with relatively small size codebooks cascaded in a three-four stage configuration. Results for log-area ratio (LAR) and line spectral pain (LSP) parameters are presented. A training technique which reduces outliers at the expense of a slight average performance degradation is introduced. The robustness across different languages and input spectral shapings is studied. Finally, it is shown that TS-MSVQ significantly outperforms the split-codebook approach.>
Bhaskar Bhattacharya, Wilf P. LeBlanc, Samy A. Mahmoud, Vladimir Cuperman
ICASSP2