Shozo Makino

According to our database1, Shozo Makino authored at least 78 papers between 1976 and 2014.

Collaborative distances:
  • Dijkstra number2 of five.
  • Erdős number3 of four.

Timeline

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Bibliography

2014
Automatic evaluation of singing enthusiasm for karaoke.
Comput. Speech Lang., 2014

2011
A System for Evaluating Singing Enthusiasm for Karaoke.
Proceedings of the 12th International Society for Music Information Retrieval Conference, 2011

Evaluation of Abnormal Sound Detection using Multi-Stage GMM in Various Environments.
Proceedings of the INTERSPEECH 2011, 2011

Bit rate reduction of the MELP coder using Lempel-Ziv segment quantization.
Proceedings of the IEEE International Conference on Acoustics, 2011

2010
Designing Side Information of Multiple Description Coding.
J. Inf. Hiding Multim. Signal Process., 2010

Speech Recognition under Multiple Noise Environment Based on Multi-Mixture HMM and Weight Optimization by the Aspect Model.
IEICE Trans. Inf. Syst., 2010

Improved Reference Speaker Weighting Using Aspect Model.
IEICE Trans. Inf. Syst., 2010

Document expansion using relevant web documents for spoken document retrieval.
Proceedings of the 6th International Conference on Natural Language Processing and Knowledge Engineering, 2010

Improvement of Packet Loss Concealment for MP3 Audio Based on Switching of Concealment Method and Estimation of MDCT Signs.
Proceedings of the Sixth International Conference on Intelligent Information Hiding and Multimedia Signal Processing (IIH-MSP 2010), 2010

Aspect-model-based reference speaker weighting.
Proceedings of the IEEE International Conference on Acoustics, 2010

2009
A speaker adaptation method for non-native speech using learners' native utterances for computer-assisted language learning systems.
Speech Commun., 2009

Novel Tonal Feature and Statistical User Modeling for Query-by-Humming.
J. Inf. Process., 2009

Automatic Query Generation and Query Relevance Measurement for Unsupervised Language Model Adaptation of Speech Recognition.
EURASIP J. Audio Speech Music. Process., 2009

Detailed description of triphone model using SSS-free algorithm.
Proceedings of the INTERSPEECH 2009, 2009

Evaluation of English intonation based on combination of multiple evaluation scores.
Proceedings of the INTERSPEECH 2009, 2009

Data Hiding is a Better Way for Transmitting Side Information for MP3 Bitstream.
Proceedings of the Fifth International Conference on Intelligent Information Hiding and Multimedia Signal Processing (IIH-MSP 2009), 2009

Detection of Abnormal Sound Using Multi-stage GMM for Surveillance Microphone.
Proceedings of the Fifth International Conference on Information Assurance and Security, 2009

2008
Multiple description coding of an audio stream by optimum recovery transforms.
J. Digit. Inf. Manag., 2008

Selection of Optimum Vocabulary and Dialog Strategy for Noise-Robust Spoken Dialog Systems.
IEICE Trans. Inf. Syst., 2008

Automatic clustering of part-of-speech for vocabulary divided PLSA language model.
Proceedings of the 4th International Conference on Natural Language Processing and Knowledge Engineering, 2008

Intonation evaluation of English utterances using synthesized speech for Computer-Assisted Language Learning.
Proceedings of the 4th International Conference on Natural Language Processing and Knowledge Engineering, 2008

Recognition of English utterances with grammatical and lexical mistakes for dialogue-based CALL system.
Proceedings of the INTERSPEECH 2008, 2008

Discrimination of task-related words for vocabulary design of spoken dialog systems.
Proceedings of the INTERSPEECH 2008, 2008

A fast speaker adaptation method using aspect model.
Proceedings of the INTERSPEECH 2008, 2008

Packet Loss Concealment for MDCT-Based Audio Codec Using Correlation-Based Side Information.
Proceedings of the 4th International Conference on Intelligent Information Hiding and Multimedia Signal Processing (IIH-MSP 2008), 2008

2007
A New Segment Quantization Using Lempel-Ziv Algorithm and Its Application to Quantization of Line Spectral Frequencies.
IEEE Trans. Commun., 2007

Music Information Retrieval from a Singing Voice Using Lyrics and Melody Information.
EURASIP J. Adv. Signal Process., 2007

Increasing Correlation using a Few Bits for Multiple Description Coding.
Proceedings of the 3rd International Conference on Intelligent Information Hiding and Multimedia Signal Processing (IIH-MSP 2007), 2007

2006
An effective music information retrieval method using three-dimensional continuous DP.
IEEE Trans. Multim., 2006

Music Information Retrieval from a Singing Voice Based on Verification of Recognized Hypotheses.
Proceedings of the ISMIR 2006, 2006

Unsupervised language model adaptation based on automatic text collection from WWW.
Proceedings of the INTERSPEECH 2006, 2006

A user simulator based on voiceXML for evaluation of spoken dialog systems.
Proceedings of the INTERSPEECH 2006, 2006

Multiple Description Coding of an Audio Stream by Optimum Recovery Transform.
Proceedings of the Second International Conference on Intelligent Information Hiding and Multimedia Signal Processing (IIH-MSP 2006), 2006

2005
Lyrics Recognition from a Singing Voice Based on Finite State Automaton for Music Information Retrieval.
Proceedings of the ISMIR 2005, 2005

Construction method of acoustic models dealing with various background noises based on combination of HMMs.
Proceedings of the INTERSPEECH 2005, 2005

Pronunciation error detection method based on error rule clustering using a decision tree.
Proceedings of the INTERSPEECH 2005, 2005

Internal noise suppression for speech recognition by small robots.
Proceedings of the INTERSPEECH 2005, 2005

A New Segment Quantizer for Line Spectral Frequencies Using Lempel-Ziv Algorithm.
Proceedings of the 2005 IEEE International Conference on Acoustics, 2005

Smile and Laughter Recognition using Speech Processing and Face Recognition from Conversation Video.
Proceedings of the 4th International Conference on Cyberworlds (CW 2005), 2005

2004
Comparison Of Features For DP-Matching Based Query-by-Humming System.
Proceedings of the ISMIR 2004, 2004

Speaker adaptation method for CALL system using bilingual speakers' utterances.
Proceedings of the INTERSPEECH 2004, 2004

A Japanese dialogue-based CALL system with mispronunciation and grammar error detection.
Proceedings of the INTERSPEECH 2004, 2004

A spoken dialog system based on automatic grammar generation and template-based weighting for autonomous mobile robots.
Proceedings of the INTERSPEECH 2004, 2004

Noise adaptive spoken dialog system based on selection of multiple dialog strategies.
Proceedings of the INTERSPEECH 2004, 2004

2003
Three-dimensional continuous DP algorithm for multiple pitch candidates in a music information retrieval system.
Proceedings of the ISMIR 2003, 2003

An optimized multi-duration HMM for spontaneous speech recognition.
Proceedings of the 8th European Conference on Speech Communication and Technology, EUROSPEECH 2003, 2003

Error Tolerant Melody Matching Method in Music Information Retrieval.
Proceedings of the Adaptive Multimedia Retrieval: First International Workshop, 2003

2002
Speech Recognition Using Acoustic Similarity-Based Primitives.
Syst. Comput. Jpn., 2002

English Speech Database Read by Japanese Learners for CALL System Development.
Proceedings of the Third International Conference on Language Resources and Evaluation, 2002

2000
Efficient segment quantization of LSP parameters for very low bit speech coding.
Proceedings of the Sixth International Conference on Spoken Language Processing, 2000

Automatic Determination Algorithm for the Optimum Number of States in NL-HMnet.
Proceedings of the Discovery Science, 2000

1999
An automatic acquisition method of statistic finite-state automaton for sentences.
Proceedings of the 1999 IEEE International Conference on Acoustics, 1999

An Automatic Acquisition of Acoustical Units for Speech Recognition Based on Hidden Markov Network.
Proceedings of the Discovery Science, 1999

1998
High-speed speaker adaptation using phoneme dependent tree-structured speaker clustering.
Proceedings of the 5th International Conference on Spoken Language Processing, Incorporating The 7th Australian International Speech Science and Technology Conference, Sydney Convention Centre, Sydney, Australia, 30th November, 1998

A study of noise robustness for speaker independent speech recognition method using phoneme similarity vector.
Proceedings of the 5th International Conference on Spoken Language Processing, Incorporating The 7th Australian International Speech Science and Technology Conference, Sydney Convention Centre, Sydney, Australia, 30th November, 1998

Automatic Acquisition of Phoneme Models and Its Application to Phoneme Labeling of a Large Size of Speech Corpus.
Proceedings of the Discovery Science, 1998

1995
A New HMnet Construction Algorithm Requiring No Contextual Factors.
IEICE Trans. Inf. Syst., 1995

Japanese document recognition based on interpolated n-gram model of character.
Proceedings of the Third International Conference on Document Analysis and Recognition, 1995

1994
Performance prediction of word recognition using the transition information between phonemes or between characters.
Syst. Comput. Jpn., 1994

A Coutinuous Speech Recognition System Using A Modified LVQ2 Method and A Dependency Grammar with Semantic Constraints.
Int. J. Pattern Recognit. Artif. Intell., 1994

Spoken word recognition using phoneme duration information estimated from speaking rate of input speech.
Proceedings of the 3rd International Conference on Spoken Language Processing, 1994

The performance prediction method on sentence recognition system using a finite state automaton.
Proceedings of ICASSP '94: IEEE International Conference on Acoustics, 1994

1993
A new word pre-selection method based on an extended redundant hash addressing for continuous speech recognition.
Proceedings of the IEEE International Conference on Acoustics, 1993

1992
Word pre-selection using a redundant hash addressing method for continuous speech recognition.
Proceedings of the Second International Conference on Spoken Language Processing, 1992

Continuous speech recognition with modified learning vector quantization algorithm and two-level DP-matching.
Proceedings of the 1992 IEEE International Conference on Acoustics, 1992

1991
A Japanese text dictation system based on phoneme recognition and a dependency grammar.
Proceedings of the 1991 International Conference on Acoustics, 1991

1990
A method of generating template patterns from a few sample patterns in character recognition.
Syst. Comput. Jpn., 1990

Performance evaluation in speech recognition system using transition probability between linguistic units.
Proceedings of the First International Conference on Spoken Language Processing, 1990

A distributed speech database with an automatic acquisition system of speech information.
Proceedings of the First International Conference on Spoken Language Processing, 1990

A Japanese text dictation system based on phoneme recognition using a modified LVQ2 method.
Proceedings of the First International Conference on Spoken Language Processing, 1990

1986
Recognition of phonemes using time-spectrum pattern.
Speech Commun., 1986

Phoneme recognition in continuous speech using phoneme discriminant filters.
Proceedings of the IEEE International Conference on Acoustics, 1986

Automatic labeling system using speaker-dependent phonetic unit references.
Proceedings of the IEEE International Conference on Acoustics, 1986

Perceptually based processing in automatic speech recognition.
Proceedings of the IEEE International Conference on Acoustics, 1986

1984
A speaker independent word recognition system based on phoneme recognition for a large size (212 words) vocabulary.
Proceedings of the IEEE International Conference on Acoustics, 1984

1983
Recognition of consonant based on the perceptron model.
Proceedings of the IEEE International Conference on Acoustics, 1983

1978
Spoken word recognition system for unlimited speakers.
Proceedings of the IEEE International Conference on Acoustics, 1978

1976
Spoken word recognition system for unlimited adult male speakers.
Proceedings of the IEEE International Conference on Acoustics, 1976


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