Demo Data from SIGIR 2026 / reviewers in the wild / expert
Dimitry Rtischev
dblp:57/9401
· DBLP profile ↗
4ranked-venue papers
1as first author
0since 2021 · last 1995
—ORCID · none
Domains — the database's venue-derived domains; a paper can count in several
Artificial intelligence and machine learning · 4 · 1 first-authorGraphics, computer vision, multimedia, augmented reality and games · 2
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
2 papers |
Speech recognition and synthesis · 100% |
Topics — the 3 heaviest of 3, each with the papers that count most for it
| Topic | Weight | Papers | Last | Evidence papers |
|---|---|---|---|---|
Natural language and speech › Speech recognition and synthesis › automatic speech recognition
speaker adaptation |
0.0 | 2 | 1995 | Speaker adaptation using constrained estimation of Gaussian mixtures · IEEE Trans. Speech Audio Process. 1995 Speaker adaptation via VQ prototype modification · IEEE Trans. Speech Audio Process. 1994 |
Natural language and speech › Speech recognition and synthesis
acoustic model adaptation |
0.0 | 1 | 1995 | Speaker adaptation using constrained estimation of Gaussian mixtures · IEEE Trans. Speech Audio Process. 1995 |
Natural language and speech › Speech recognition and synthesis
acoustic modeling |
0.0 | 1 | 1994 | Speaker adaptation via VQ prototype modification · IEEE Trans. Speech Audio Process. 1994 |
Methods — techniques the papers use, named apart from their topics
maximum likelihood estimation · 0.0vector quantization · 0.0tied-mixture continuous-parameter HMM · 0.0
| Year | Publication | Venue | Position |
|---|---|---|---|
| 1995 | Speaker adaptation using constrained estimation of Gaussian mixturesabstractA trend in automatic speech recognition systems is the use of continuous mixture-density hidden Markov models (HMMs). Despite the good recognition performance that these systems achieve on average in large vocabulary applications, there is a large variability in performance across speakers. Performance degrades dramatically when the user is radically different from the training population. A popular technique that can improve the performance and robustness of a speech recognition system is adapting speech models to the speaker, and more generally to the channel and the task. In continuous mixture-density HMMs the number of component densities is typically very large, and it may not be feasible to acquire a sufficient amount of adaptation data for robust maximum-likelihood estimates. To solve this problem, the authors propose a constrained estimation technique for Gaussian mixture densities. The algorithm is evaluated on the large-vocabulary Wall Street Journal corpus for both native and nonnative speakers of American English. For nonnative speakers, the recognition error rate is approximately halved with only a small amount of adaptation data, and it approaches the speaker-independent accuracy achieved for native speakers. For native speakers, the recognition performance after adaptation improves to the accuracy of speaker-dependent systems that use six times as much training data.> Vassilios Digalakis, Dimitry Rtischev, Leonardo Neumeyer |
IEEE Trans. Speech Audio Process. | 2 |
| 1994 | Speaker adaptation via VQ prototype modificationabstractA statistical technique for vector quantizer (VQ) prototype adaptation, based on tied-mixture continuous-parameter HMM's, is derived and evaluated on the basis of experimental evidence. Performance on difficult adaptation tasks indicates that VQ-prototype adaptation via tied-mixture HMM's constitutes a useful mechanism for speaker adaptation, particularly when there are substantial channel differences or when there is a large mismatch between reference and target speaker characteristics. Dimitry Rtischev, David Nahamoo, Michael Picheny |
IEEE Trans. Speech Audio Process. | 1 |
| 1991 | A voice interactive language instruction system
Jared Bernstein, Dimitry Rtischev |
EUROSPEECH | 2 |
| 1990 | Automatic evaluation and training in English pronunciationabstractSRI is developing a system that uses real time speech recognition to diag nose, evaluate and provide training in spoken English. The paper first describes the methods and results of a study of the feasibility of automati cally grading the performance of Japanese students when reading English aloud. Utterances recorded from Japanese speakers were independently rated by expert listeners. Speech grading software was developed from a speaker independent hidden-Markov-model speech recognition system. The auto matic grading procedure first aligned the speech with a model and then com pared the segments of the speech signal with models of those segments that have been developed from a database of speech from native speakers of English. The evaluation study showed that ratings of speech quality by experts are very reliable and automatic grades correlate well (r > 0.8) with those expert ratings. SRI is now extending this technology and integrating it in a spoken-language training system. This effort involves (1) porting SRI's DECIPHER speech recognition system to a microcomputer platform, and (2) extending the speech-evaluation software to more exactly diagnose a learner's pronuncia tion deficits and lead the learner through an appropriate regimen of exer cises. Jared Bernstein, Hy Murveit, Dimitry Rtischev, Mitch Weintraub |
ICSLP | 4 |