• Journal of Korea Multimedia Society
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• v.6 no.3
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• pp.462-468
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• 2003

In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech quality in case coexist with a voiced and an unvoiced consonants in a frame. So, I propose TSIUVC(Transition Segment Including Unvoiced Consonant) searching and extraction method in order to uncoexistent with a voiced and unvoiced consonants in a frame. This paper present a new method of TSIUVC approximate-synthesis by using Least Mean Square and frequency band division. As a result, this method obtain a high quality approximation-synthesis waveforms within TSIUVC by using frequency information of 0.547KHz below and 2.813KHz above. The important thing is that the maximum error signal can be made with low distortion approximation-synthesis waveform within TSIUVC. This method has the capability of being applied to a new speech coding of Voiced/Silence/TSIUVC, speech analysis and speech synthesis.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.12 no.9
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• pp.4119-4124
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• 2011

In a MPC coding using excitation source of voiced and unvoiced, it would be a distortion of speech waveform in case of increasing or decreasing of speech signal amplitude in a frame. This is caused by normalization of synthesis speech signal in the process of restoration the multi-pulses of representation section. To solve this problem, this paper present a method of amplitude compensation(AC-MPC) in a multi-pulses each pitch interval in order to reduce distortion of speech waveform. I was confirmed that the method can be synthesized close to the original speech waveform. And I evaluate the MPC and AC-MPC using amplitude compensation method. As a result, SNRseg of AC-MPC was improved 0.7dB for female voice and 0.7dB for male voice respectively. Compared to the MPC, SNRseg of AC-MPC has been improved that I was able to control the distortion of the speech waveform finally. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.17 no.7
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• pp.49-53
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• 2016

Recently, the use of signal compression methods to improve the efficiency of wireless networks have increased. In particular, the MPC system was used in the pitch extraction method and the excitation source of voiced and unvoiced to reduce the bit rate. In general, the MPC system using an excitation source of voiced and unvoiced would result in a distortion of the synthesis speech waveform in the case of voiced and unvoiced consonants in a frame. This is caused by normalization of the synthesis speech waveform in the process of restoring the multi-pulses of the representation segment. This paper presents an ACFBD-MPC (Amplitude Compensation Frequency Band Division-Multi Pulse Coding) using amplitude compensation in a multi-pulses each pitch interval and specific frequency to reduce the distortion of the synthesis speech waveform. The experiments were performed with 16 sentences of male and female voices. The voice signal was A/D converted to 10kHz 12bit. In addition, the ACFBD-MPC system was realized and the SNR of the ACFBD-MPC estimated in the coding condition of 8kbps. As a result, the SNR of ACFBD-MPC was 13.6dB for the female voice and 14.2dB for the male voice. The ACFBD-MPC improved the male and female voice by 1 dB and 0.9 dB, respectively, compared to the traditional MPC. This method is expected to be used for cellular telephones and smartphones using the excitation source with a low bit rate.

• Journal of Internet Computing and Services
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• v.18 no.5
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• pp.17-22
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• 2017

In a MPC coding using excitation source of voiced and unvoiced, it would be a distortion of speech waveform. This is caused by normalization of synthesis speech waveform of voiced in the process of restoration the multi-pulses of representation section. This paper present PCFBD-MPC( Position Compensation Frequency Band Division-Multi Pulse Coding ) used V/UV/S( Voiced / Unvoiced / Silence ) switching, position compensation in a multi-pulses each pitch interval and Unvoiced approximate-synthesis by using specific frequency in order to reduce distortion of synthesis waveform. Also, I was implemented that the PCFBD-MPC( Position Compensation Frequency Band Division-Multi Pulse Coding ) system and evaluate the SNRseg of PCFBD-MPC in coding condition of 8kbps. As a result, SNRseg of PCFBD-MPC was 13.4dB for female voice and 13.8dB for male voice respectively. In the future, I will study the evaluation of the sound quality of 8kbps speech coding method that simultaneously compensation the amplitude and position of multi-pulse source. These methods are expected to be applied to a method of speech coding using sound source in a low bit rate such as a cellular phone or a smart phone.

• The Journal of the Korea Contents Association
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• v.5 no.3
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• pp.167-173
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• 2005

In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech quality in case coexist with a voiced and unvoiced consonants in a frame. So, I propose TSIUVC(Transition Segment Including Unvoiced Consonant) extraction method by using pitch pulses and Zero Crossing Rate in order to unexistent with a voiced and unvoiced consonants in a frame. And this paper present a TSIUVC approximate-synthesis method by using frequency band division. As a result, this method obtains a high quality approximation-synthesis waveform within TSIUVC by using frequency information of 0.547kHz below and 2.813kHz above. And the TSIUVC extraction rate was $91\%$ for female voice and $96.2\%$ for male voice respectively This method has the capability of being applied to a new speech coding of Voiced/Silence/TSIUVC, speech analysis, and speech synthesis.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1247-1250
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• 1998

In speech synthesis, LF model is widely used for excitation signal for voice source coding system. But LF model does not represent the harmonic frequencies of excitation signal. We propose an effective method which use sinusoidal functions for representing the harmonics of voice source signal. The proposed method could achieve more exact voice source waveform and better synthesized speech quality than LF model.

• Journal of the Korean Institute of Telematics and Electronics
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• v.25 no.2
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• pp.176-181
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• 1988

In this paper, the LPC speech synthesizer with Multipulsse excitation is implemented using general-purpose DSP \ulcornerD7720. As the driving function for synthesis filter is used in the amplitude and position of pulse, the Voice/Unvoice decision and pitch period detectioncan be excluded. The synthesizer is implemented with DSP device which is operated on the interrupt mehtod with main computer and on the DMA mehtod with D/A converter. The comparision of synthetic and original waveform, alogn with the listening test, proves the validity of this system.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.10 no.4
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• pp.159-167
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• 1985

It was demonstrated that the glottal waves are different depending on a kind of vowels in deriving the glottal waves directly from Korean vowels/a, e, I, o, u/ w, ch are recorded by a male speaker. After resynthesizing vowels with five simulated glottal waves, the effects of glottal wave shape on the speech synthesis were compared with in terms of waveform. Some changes could be seen in the waveforms of the synthetic vowels with the variation of the shape, opening time and closing time, therefore it was confirmed that in the speech sysnthesis, the glottal wave shape is an important factor in the improvement of the speech quality.

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.410-413
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• 2000

This paper suggests a methodology in which control signals for high-level synthesis are generated from SDL specification. SDL is based on EFSM(Extended Finite State Machine) model. Data path and control part are partitioned into representing data operations in the from of scheduled data flow graph and process behavior of an SDL code in forms of an abstract FSM. Resource allocation is performed based on the suggested architecture model and local control signals to drive allocated functional blocks are incorporated into an abstract FSM extracted from an SDL process specification. Data path and global controller acquired through suggested methodology are combined into structural VHDL representation and correctness of behavior for final circuit is verified through waveform simulation.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.763-767
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• 2008

In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech qualify in case coexist with a voiced and an unvoiced consonants in a frame. So, We proposed TS(Transition Segment) including unvoiced consonant searching and extraction method in order to uncoexistent with a voiced and unvoiced consonants in a frame. This research present a new method of TS approximate-synthesis by using Least Mean Square and frequency band division. As a result, this method obtain a high quality approximation-synthesis waveforms within TS by using frequency information of 0.547kHz below and 2.813kHz above. The important thing is that the maximum error signal can be made with low distortion approximation-synthesis waveform within TS. This method has the capability of being applied to a new speech coding of Voiced/Silence/TS, speech analysis and speech synthesis.