• Title/Summary/Keyword: sound recognition algorithm

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QRAS-based Algorithm for Omnidirectional Sound Source Determination Without Blind Spots (사각영역이 없는 전방향 음원인식을 위한 QRAS 기반의 알고리즘)

  • Kim, Youngeon;Park, Gooman
    • Journal of Broadcast Engineering
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    • v.27 no.1
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    • pp.91-103
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    • 2022
  • Determination of sound source characteristics such as: sound volume, direction and distance to the source is one of the important techniques for unmanned systems like autonomous vehicles, robot systems and AI speakers. There are multiple methods of determining the direction and distance to the sound source, e.g., using a radar, a rider, an ultrasonic wave and a RF signal with a sound. These methods require the transmission of signals and cannot accurately identify sound sources generated in the obstructed region due to obstacles. In this paper, we have implemented and evaluated a method of detecting and identifying the sound in the audible frequency band by a method of recognizing the volume, direction, and distance to the sound source that is generated in the periphery including the invisible region. A cross-shaped based sound source recognition algorithm, which is mainly used for identifying a sound source, can measure the volume and locate the direction of the sound source, but the method has a problem with "blind spots". In addition, a serious limitation for this type of algorithm is lack of capability to determine the distance to the sound source. In order to overcome the limitations of this existing method, we propose a QRAS-based algorithm that uses rectangular-shaped technology. This method can determine the volume, direction, and distance to the sound source, which is an improvement over the cross-shaped based algorithm. The QRAS-based algorithm for the OSSD uses 6 AITDs derived from four microphones which are deployed in a rectangular-shaped configuration. The QRAS-based algorithm can solve existing problems of the cross-shaped based algorithms like blind spots, and it can determine the distance to the sound source. Experiments have demonstrated that the proposed QRAS-based algorithm for OSSD can reliably determine sound volume along with direction and distance to the sound source, which avoiding blind spots.

A Study on Phoneme Likely Units to Improve the Performance of Context-dependent Acoustic Models in Speech Recognition (음성인식에서 문맥의존 음향모델의 성능향상을 위한 유사음소단위에 관한 연구)

  • 임영춘;오세진;김광동;노덕규;송민규;정현열
    • The Journal of the Acoustical Society of Korea
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    • v.22 no.5
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    • pp.388-402
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    • 2003
  • In this paper, we carried out the word, 4 continuous digits. continuous, and task-independent word recognition experiments to verify the effectiveness of the re-defined phoneme-likely units (PLUs) for the phonetic decision tree based HM-Net (Hidden Markov Network) context-dependent (CD) acoustic modeling in Korean appropriately. In case of the 48 PLUs, the phonemes /ㅂ/, /ㄷ/, /ㄱ/ are separated by initial sound, medial vowel, final consonant, and the consonants /ㄹ/, /ㅈ/, /ㅎ/ are also separated by initial sound, final consonant according to the position of syllable, word, and sentence, respectively. In this paper. therefore, we re-define the 39 PLUs by unifying the one phoneme in the separated initial sound, medial vowel, and final consonant of the 48 PLUs to construct the CD acoustic models effectively. Through the experimental results using the re-defined 39 PLUs, in word recognition experiments with the context-independent (CI) acoustic models, the 48 PLUs has an average of 7.06%, higher recognition accuracy than the 39 PLUs used. But in the speaker-independent word recognition experiments with the CD acoustic models, the 39 PLUs has an average of 0.61% better recognition accuracy than the 48 PLUs used. In the 4 continuous digits recognition experiments with the liaison phenomena. the 39 PLUs has also an average of 6.55% higher recognition accuracy. And then, in continuous speech recognition experiments, the 39 PLUs has an average of 15.08% better recognition accuracy than the 48 PLUs used too. Finally, though the 48, 39 PLUs have the lower recognition accuracy, the 39 PLUs has an average of 1.17% higher recognition characteristic than the 48 PLUs used in the task-independent word recognition experiments according to the unknown contextual factor. Through the above experiments, we verified the effectiveness of the re-defined 39 PLUs compared to the 48PLUs to construct the CD acoustic models in this paper.

Modification of Pitch Algorithm and Its Application to Noise (피치 알고리즘 수정 및 소음에의 적용)

  • Shin, Sung-Hwan;Ih, Jeong-Guon
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2002.11a
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    • pp.354.1-354
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    • 2002
  • Pitch is a perception related to frequency, one of the psychological aspects or attributes of tones, and an important factor to determine sound quality of sound together with loudness and timber. while a study on pitch has been actively achieved In the part of speech recognition and speech separation, that for analysis and improvement of product sound quality is not yet enough. (omitted)

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A Merging Algorithm with the Discrete Wavelet Transform to Extract Valid Speech-Sounds (이산 웨이브렛 변환을 이용한 유효 음성 추출을 위한 머징 알고리즘)

  • Kim, Jin-Ok;Hwang, Dae-Jun;Paek, Han-Wook;Chung, Chin-Hyun
    • Journal of KIISE:Computing Practices and Letters
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    • v.8 no.3
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    • pp.289-294
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    • 2002
  • A valid speech-sound block can be classified to provide important information for speech recognition. The classification of the speech-sound block comes from the MRA(multi-resolution analysis) property of the DWT(discrete wavelet transform), which is used to reduce the computational time for the pre-processing of speech recognition. The merging algorithm is proposed to extract valid speech-sounds in terms of position and frequency range. It needs some numerical methods for an adaptive DWT implementation and performs unvoiced/voiced classification and denoising. Since the merging algorithm can decide the processing parameters relating to voices only and is independent of system noises, it is useful for extracting valid speech-sounds. The merging algorithm has an adaptive feature for arbitrary system noises and an excellent denoising SNR(signal-to-nolle ratio).

Development of a Collapse-sensing Phone and Collapse Recognition Algorithm (낙상 감지 폰의 개발과 낙상판단 알고리즘)

  • Jang, Duk-Sung
    • IEMEK Journal of Embedded Systems and Applications
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    • v.10 no.1
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    • pp.41-48
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    • 2015
  • To deal with the emergency of the solitary aged people, we have developed a collapse-sensing phone, in which a collapse sensor, a GPS receiving chipset and a CDMA sending chipset are included. The general cellular phone is somewhat expensive communication device using sound and characters, but the collapse-sensing phone is a cheaper and popular version. If the collapse sensor recognizes a certain of collapse of the aged people, CDMA sending chipset will send the location of the phone which is received from satellite by GPS receiving chipset. In this paper, a collapse recognition algorithm which is developed by using much experimental data, will be introduced to explain how to recognize the real collapse from fast sitting or immediate standing after collapse. Once a true collapse is ecognized, the phone-ID and the coordinate will be sent to the server of administrative office via CDMA network. And the position of emergency will be displayed on the GIS with the rescue center.

Convolutional Neural Network Based Image Processing System

  • Kim, Hankil;Kim, Jinyoung;Jung, Hoekyung
    • Journal of information and communication convergence engineering
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    • v.16 no.3
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    • pp.160-165
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    • 2018
  • This paper designed and developed the image processing system of integrating feature extraction and matching by using convolutional neural network (CNN), rather than relying on the simple method of processing feature extraction and matching separately in the image processing of conventional image recognition system. To implement it, the proposed system enables CNN to operate and analyze the performance of conventional image processing system. This system extracts the features of an image using CNN and then learns them by the neural network. The proposed system showed 84% accuracy of recognition. The proposed system is a model of recognizing learned images by deep learning. Therefore, it can run in batch and work easily under any platform (including embedded platform) that can read all kinds of files anytime. Also, it does not require the implementing of feature extraction algorithm and matching algorithm therefore it can save time and it is efficient. As a result, it can be widely used as an image recognition program.

Development of a Multiple Monitioring System for Intelligence of a Machine Tool -Application to Drilling Process- (공작기계 지능화를 위한 다중 감시 시스템의 개발-드릴가공에의 적용-)

  • Kim, H.Y.;Ahn, J.H.
    • Journal of the Korean Society for Precision Engineering
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    • v.10 no.4
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    • pp.142-151
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    • 1993
  • An intelligent mulitiple monitoring system to monitor tool/machining states synthetically was proposed and developed. It consists of 2 fundamental subsystems : the multiple sensor detection unit and the intellignet integrated diagnosis unit. Three signals, that is, spindle motor current, Z-axis motor current, and machining sound were adopted to detect tool/machining states more reliably. Based on the multiple sensor information, the diagnosis unit judges either tool breakage or degree of tool wear state using fuzzy reasoning. Tool breakage is diagnosed by the level of spindle/z-axis motor current. Tool wear is diagnosed by both the result of fuzzy pattern recognition for motor currents and the result of pattern matching for machining sound. Fuzzy c-means algorithm was used for fuzzy pattern recognition. Experiments carried out for drill operation in the machining center have shown that the developed system monitors abnormal drill/states drilling very reliably.

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Ortho-phonic Alphabet Creation by the Musical Theory and its Segmental Algorithm (악리론으로 본 정음창제와 정음소 분절 알고리즘)

  • Chin, Yong-Ohk;Ahn, Cheong-Keung
    • Speech Sciences
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    • v.8 no.2
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    • pp.49-59
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    • 2001
  • The phoneme segmentation is a very difficult problem in speech sound processing because it has found out segmental algorithm in many kinds of allophone and coarticulation's trees. Thus system configuration for the speech recognition and voice retrieval processing has a complex system structure. To solve it, we discuss a possibility of new segmental algorithm, which is called the minus a thirds one or plus in tripartitioning(삼분손익) of twelve temporament(12 율려), first proposed by Prof. T. S. Han. It is close to oriental and western musical theory. He also has suggested a 3 consonant and 3 vowel phonemes in Hunminjungum(훈민정음) invented by the King Sejong in the 15th century. In this paper, we suggest to newly name it as ortho-phonic phoneme(OPP/정음소), which carries the meaning of 'the absoluteness and independency'. OPP also is acceptable to any other languages, for example IPA. Lastly we know that this algorithm is constantly applicable to the global language and is very useful to construct a voice recognition and retrieval structuring engineering.

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Study on the Self Diagnostic Monitoring System for an Air-Operated Valve : Algorithm for Diagnosing Defects

  • Kim Wooshik;Chai Jangbom;Choi Hyunwoo
    • Nuclear Engineering and Technology
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    • v.36 no.3
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    • pp.219-228
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    • 2004
  • [1] and [2] present an approach to diagnosing possible defects in the mechanical systems of a nuclear power plant. In this paper, by using a fault library as a database and training data, we develop a diagnostic algorithm 1) to decide whether an Air Operated Valve system is sound or not and 2) to identify the defect from which an Air-Operated Valve system suffers, if any. This algorithm is composed of three stages: a neural net stage, a non-neural net stage, and an integration stage. The neural net stage is a simple perceptron, a pattern-recognition module, using a neural net. The non-neural net stage is a simple pattern-matching algorithm, which translates the degree of matching into a corresponding number. The integration stage collects each output and makes a decision. We present a simulation result and confirm that the developed algorithm works accurately, if the input matches one in the database.

Speaker Adapted Real-time Dialogue Speech Recognition Considering Korean Vocal Sound System (한국어 음운체계를 고려한 화자적응 실시간 단모음인식에 관한 연구)

  • Hwang, Seon-Min;Yun, Han-Kyung;Song, Bok-Hee
    • The Journal of Korea Institute of Information, Electronics, and Communication Technology
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    • v.6 no.4
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    • pp.201-207
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    • 2013
  • Voice Recognition technique has been developed and it has been actively applied to various information devices such as smart phones and car navigation system. But the basic research technique related the speech recognition is based on research results in English. Since the lip sync producing generally requires tedious hand work of animators and it serious affects the animation producing cost and development period to get a high quality lip animation. In this research, a real time processed automatic lip sync algorithm for virtual characters in digital contents is studied by considering Korean vocal sound system. This suggested algorithm contributes to produce a natural lip animation with the lower producing cost and the shorter development period.