Demo Data from EDBT 2026 / reviewers in the wild / expert
Alv I. Aarskog
dblp:74/11421
· DBLP profile ↗
2ranked-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 · 1 first-author
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.
| Computer graphics and multimedia
1 paper |
Audio and music processing · 70% Image and video coding · 30% |
Topics — the 4 heaviest of 4, each with the papers that count most for it
| Topic | Weight | Papers | Last | Evidence papers |
|---|---|---|---|---|
Audio and music processing › speech coding
linear predictive coding |
0.0 | 1 | 1994 | Predictive coding of speech using microphone/speaker adaptation and vector quantization · IEEE Trans. Speech Audio Process. 1994 |
Audio and music processing
speech coding |
0.0 | 1 | 1994 | Predictive coding of speech using microphone/speaker adaptation and vector quantization · IEEE Trans. Speech Audio Process. 1994 |
Image and video coding › quantization
vector quantization |
0.0 | 1 | 1994 | Predictive coding of speech using microphone/speaker adaptation and vector quantization · IEEE Trans. Speech Audio Process. 1994 |
Audio and music processing › speech coding
code-excited linear prediction |
0.0 | 1 | 1994 | Predictive coding of speech using microphone/speaker adaptation and vector quantization · IEEE Trans. Speech Audio Process. 1994 |
Methods — techniques the papers use, named apart from their topics
microphone/speaker adaptation filter · 0.0backward adaptive prefiltering · 0.0
| Year | Publication | Venue | Position |
|---|---|---|---|
| 1994 | Predictive coding of speech using microphone/speaker adaptation and vector quantizationabstractA problem with speech coders based on trained quantizers is their lack of robustness against variations in the microphone and input filter response. In this paper a simple backward adaptive prefiltering technique is proposed as a means of improving the robustness and quality of a speech coder at no cost in bit rate. The technique is particularly useful in conjunction with vector quantization (VQ) of the linear predictive coding (LPC) parameters. The performance of the prefilter, denoted a microphone and speaker adaptation (MSA) filter, has been evaluated in terms of prediction gain and spectral distortion, together with objective and subjective quality of a 7.5 kbit/s CELP speech coder. In this coder a 10-bit direct VQ of the LPC parameters using the residual energy distortion measure has been applied. This is consistent with the covariance method of LPC analysis. Simulation results illustrate that the MSA filter significantly improves the performance and robustness of the LPC VQ against changes in the input response. The 7.5 kbit/s CELP with a trained excitation codebook and MSA was found to be clearly better (subjectively and objectively) than the one without MSA. The coder with MSA also showed to be practically indistinguishable from the same CELP with unquantized LPC coefficients and a stochastic excitation codebook.> Alv I. Aarskog, Hans C. Guren |
IEEE Trans. Speech Audio Process. | 1 |
| 1991 | A long-term predictive ADPCM coder with short-term prediction and vector quantizationabstractA speech coder structure based on the long-term predictive ADPCM (adaptive differential pulse-code modulation) coder with vector quantization of pitch coefficients, the LPC (linear predictive coding) coefficients, and the residual signal is presented. In order to improve the robustness against variations in the average spectral envelope a backward adaptive microphone and speaker adaptation filter is used for preprocessing the speech. With this coder, a good delay/quality tradeoff can be obtained. Simulations of a 8.133 kb/s coder with an algorithmic delay of only 7.5 ms show that a very good speech quality is produced.> Alv I. Aarskog, Arild Nilsen, Ole Berg, Hans C. Guren |
ICASSP | 1 |