• ETRI Journal
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• v.36 no.1
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• pp.175-178
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• 2014

We introduce the Energy-Based Blind Separating (EBS) algorithm for extremely fast separation of mixed speech signals without loss of quality, which is performed in two stages: iterative-form separation and closed-form separation. This algorithm significantly improves the separation speed simply due to incorporating only some specific frequency bins into computations. Simulation results show that, on average, the proposed algorithm is 43 times faster than the independent component analysis (ICA) for speech signals, while preserving the separation quality. Also, it outperforms the fast independent component analysis (FastICA), the joint approximate diagonalization of eigenmatrices (JADE), and the second-order blind identification (SOBI) algorithm in terms of separation quality.

• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.1131-1134
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• 1999

In general CELP type vocoders provide good speech quality around 4.8kbps. Among them, G.723.1 developed for Internet Phone and videoconferencing includes two vocoders, 5.3kbps ACELP and 6.3kbps MO-MLQ. Since 6.3kbps MP-MLQ requires large amount of computation for fixed codebook search, it is difficult to realize real time processing. In order to improve the problem this paper proposes the new method that reduces the processing time up to about 50% of codebook search time. We first decide the grid bit, then search the codebook. Grid bit is selected by comparison between synthetic speech, which is synthesized with only odd or even pulses of target vector. and DC removed original speech. As a result, we reduced the total processing time of G.723.1 MP-MLQ up to about 26.08%. In objective quality test 11.19㏈ of segSNR was obtained, and in subjective quality test there was almost no speech degradation.

• Proceedings of the Acoustical Society of Korea Conference
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• 1991.06a
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• pp.67-72
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• 1991

This paper evaluates thequality of synthesized speech by rule using the LPC CD in the objective measure and then compares the result with the subjective analysis. By evaluating the quality of synthesized speech by rule objectively. We have tried to resolve the problems (Evaluation time or size expansion, variables within the analysis results) that arise when the evaluation is done subjectively. Also by comparing intelligibility-the index for the subjective quality evaluation of synthesized speech by rule-with evaluation results obtained using MOS and the objective evaluation. We have proved the validity of the objective analysis and thus provides a guide that would be useful when R&D and marketing of synthesis by rule method is done.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.29B no.1
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• pp.43-49
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• 1992

As a result of recent progress in speech synthesis techniques, the those new services using new techniques are going to introduce into the telephone communication system. In setting standards, voice quality is obviously an important criterion. It is very important to develope a quality evaluation method of synthesized speech for the diagnostic assessment of system algorithm, and fair comparison of assessment values. This paper has described several basic concepts and criterions for quality assessment (intelligibility) of synthesized speech by rule, and then a word selection method and the word list to be used in word intelligibility test were proposed. Finally, a test method for word intelligibility is described.

• ETRI Journal
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• v.31 no.4
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• pp.408-418
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• 2009

Adopting an encryption function in voice over Wi-Fi service incurs problems such as additional power consumption and degradation of communication quality. To overcome these problems, a partial encryption (PE) algorithm for compressed speech was recently introduced. However, from the security point of view, the partial encryption sets (PESs) of the conventional PE algorithm still have much room for improvement. This paper proposes a new selection method for finding a smaller PES while maintaining the security level of encrypted speech. The proposed PES selection method employs the perceptual evaluation of the speech quality (PESQ) algorithm to objectively measure the distortion of speech. The proposed method is applied to the ITU-T G.729 speech codec, and content protection capability is verified by a range of tests and a reconstruction attack. The experimental results show that encrypting only 20% of the compressed bitstream is sufficient to effectively hide the entire content of speech.

• ETRI Journal
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• v.31 no.4
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• pp.457-459
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• 2009

The optimum maximum voiced frequency (MVF) estimation-based two-band excitation for hidden Markov model-based speech synthesis is presented. An analysis-by-synthesis scheme is adopted for the MVF estimation which leads to the minimum spectral distortion of synthesized speech. Experimental results show that the proposed method significantly improves synthetic speech quality.

• MALSORI
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• no.43
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• pp.89-99
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• 2002

Recently, researches into ways of improving large vocabulary continuous speech recognition and speech synthesis are being carried out intensively as the field of speech information technology is progressing rapidly. In the field of speech recognition, developments of stochastic methods such as HMM require large amount of speech data for training, and also in the field of speech synthesis, recent practices show that synthesis of better quality can be produced by selecting and connecting only the variable size of speech data from the large amount of speech data. In this paper we design and discuss linguistic contents for speech copora for speech recognition and synthesis to be shared in common.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.699-707
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• 1999

It cannot be argued that speech is the most natural interfacing tool between men and machines. In order to realize acceptable speech interfaces, highly advanced speech recognizers and synthesizers are inevitable. Text-to-Speech(TTS) technology has been attracting a lot of interest among speech engineers because of its own benefits. Namely, the possible application areas of talking computers, emergency alarming systems in speech, speech output devices fur speech-impaired, and so on. Hence, many researchers have made significant progresses in the speech synthesis techniques in the sense of their own languages and as a result, the quality of currently available speech synthesizers are believed to be acceptable to normal users. These are partly why the MPEG group had decided to include the TTS technology as one of its MPEG-4 functionalities. ETRI has made major contributions to the current MPEG-4 TTS among various MPEG-4 functionalities. They are; 1) use of original prosody for synthesized speech output, 2) trick mode functions fer general users without breaking synthesized speech prosody, 3) interoperability with Facial Animation(FA) tools, and 4) dubbing a moving/animated picture with lib-shape pattern information.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.10 no.4
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• pp.71-77
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• 2011

In this paper, we propose a conversational quality measurement (CQM) system for providing the objective QoS of high quality mobile VoIP voice telecommunication. For measuring the conversational quality, the VoIP telecommunication system is implemented in two smart phones connected with VoIP. The VoIP telecommunication system consists of echo cancellation, noise reduction, speech encoding/decoding, packet generation with RTP (Real-Time Protocol), jitter buffer control and POS (Play-out Schedule) with LC (loss Concealment). The CQM system is connected to a microphone and a speaker of each smart phone. The voice signal of each speaker is recorded and used to measure CE (Conversational Efficiency), CS (Conversational Symmetry), PESQ (Perceptual Evaluation of Speech Quality) and CE-CS-PESQ correlation. We prove the CQM system by measuring CE, CS and PESQ under various SNR, delay and loss due to IP network environment.

• Journal of the Korea Society of Computer and Information
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• v.29 no.1
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• pp.41-49
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• 2024

Speech recognition technology is continuously advancing and widely used in various fields. In this study, we aimed to investigate the impact of speech data quality on speech recognition models by dividing the dataset into the entire dataset and the top 70% based on Signal-to-Noise Ratio (SNR). Utilizing Seamless M4T and Google Cloud Speech-to-Text, we examined the text transformation results for each model and evaluated them using the Levenshtein Distance. Experimental results revealed that Seamless M4T scored 13.6 in models using data with high SNR, which is lower than the score of 16.6 for the entire dataset. However, Google Cloud Speech-to-Text scored 8.3 on the entire dataset, indicating lower performance than data with high SNR. This suggests that using data with high SNR during the training of a new speech recognition model can have an impact, and Levenshtein Distance can serve as a metric for evaluating speech recognition models.