• Speech Sciences
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• v.8 no.4
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• pp.35-43
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• 2001

New speech synthesis algorithms capable of flexible prosody (especially F0) modification are desired for a high quality TTS system. TD-PSOLA is the most popular synthesis algorithm. The algorithm shows very high quality when F0 modification is limited. However, the quality degradation due to pitch epoch detection error becomes severe as the F0 modification factor becomes large. On the other hand, the vocoder framework is very flexible in F0 manipulation. The synthesized speech quality from the vocoder is far from natural human speech and suffers from buzziness. To remedy the buzzy quality from the vocoder and make more natural synthetic speech, we propose a mixed phase vocoder.

• The Journal of the Acoustical Society of Korea
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• v.1 no.1
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• pp.54-59
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• 1982

We present an improved 4800 bps LPC vocoder system that virtually eleminates the buzzy effect from synthetic speech. Excitation signal in the new system is formed by adding high-pass filtered pitch pulses or random noise to a baseband residual signal that has been coded by pitch predictive PCM. Since the baseband residual is used as a part of excitation, the system is also robust to V/UV and pitch errors. According to our informal listening tests, the synthetic speech of the new system does not have the buzzy effect. As a result the vocoder speech quality is more natural than that of a conventioinal LPC vocoder.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.106-109
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• 2000

There have been proposed two types of low bit rate vocoder upto now : One is MBE type using the spectrum modeling and another is CELP type using the hybrid coding method. CELP type vocoder has mainly studied between them. Specially, much of intensity is concentrated in CELP vocoder due to the emergence of Internet Phone and PCS in a domestic. In order to improve the speech quality in CELP vocoder, in this paper, we proposed a new spectrum analysis algorithm with variable window, In CELP vocoder, the spectrum of the synthesised speech signal is distorted because the fixed size windows is used for spectrum analysis. So we have measured the spectral leakage and in order to minimize the spectral leakage have adjusted the window size. Applying this method G.723.1 ACELP, we can get SD(Spectral Distortion) reduction 0.084(dB), residual energy reduction 6.3% and MOS(Mean Opinion Score) improvement 0.1.

• Proceedings of the KAIS Fall Conference
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• 2010.05a
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• pp.265-268
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• 2010

There have been proposed two types of low bit rate vocoder upto now : One is MBE type using the spectrum modeling and another is CELP type using the hybrid coding method. CELP type vocoder has mainly studied between them. Specially, much of intensity is concentrated in CELP vocoder due to the emergence of Internet Phone and PCS in a domestic. In order to improve the speech quality in CELP vocoder, in this paper, we proposed a new spectrum analysis algorithm with variable window. In CELP vocoder, the spectrum of the synthesised speech signal is distorted because the fixed size windows is used for spectrum analysis. So we have measured the spectral leakage and in order to minimize the spectral leakage have adjusted the window size. Applying this method G.723.1 ACELP, we can get SD(Spectral Distortion) reduction 0.084(dB), residual energy reduction 6.3% and MOS(Mean Opinion Score) improvement 0.1.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.14 no.6
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• pp.699-712
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• 1989

This paper describes a method of Korean speech synhes is using format VOCODER. The parameters of speech synthes is are a follows, 1) format F1, F2, and F3 by spectrum moment method and F4, F5 using the length of vocal tract. 2) pitch frequencies obtained by optimu, Comb method using AMDF. 3) short time average energy and short time mean amplitude. 4) The decision method of bandwidth reportd by Fant. 5) voicde/unvoiced discrimination using zerocrossing. 6) excitation wave reported by Rosenberg. 7) gaussian white noise. Synthesis results are in fairly good agreement with original speech.

• Speech Sciences
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• v.10 no.3
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• pp.241-250
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• 2003

In this paper, we implemented the vocoder of variable rate by applying the SOLA-B algorithm to the G.729A to the TMS320C5416 in real-time. This method using the SOLA-B algorithm is that it is reduced the duration of the speech in encoding and is played at the speed of normal by extending the duration of the speech in decoding. But the method applied to the existed G.729A and SOLA-B algorithm is caused the loss of speech quality in G.729A which is not reflected about length variation of speech. Therefore the proposed method is encoded according as it is modified the structure of LSP quantization table about the length of speech is reduced by using the SOLA-B algorithm. The vocoder of variable rate by applying the G.729A and SOLA-B algorithm is represented the maximum complexity of 10.2MIPS about encoder and 2.8MIPS about decoder in 8kbps transmission rate. Also it is evaluated 17.3MIPS about encoder, 9.9MIPS about decoder in 6kbps and 18.5MIPS about encoder, 11.1MIPS about decoder in 4kbps according to the transmission rate. The used memory is about program ROM 9.7kwords, table ROM 4.69kwords, RAM 5.2kwords. The waveform of output is showed by the result of C simulator and Bit Exact. Also, the result of MOS test for evaluation of speech quality of the vocoder of variable rate which is implemented in real-time, it is estimated about 3.68 in 4kbps.

• Speech Sciences
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• v.10 no.4
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• pp.125-136
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• 2003

Speech coding techniques have been studied to reduce the complexity and bit rate but also to improve the sound quality. CELP type vocoder, has used as a one of standard, supports the great sound quality even low bit rate. In this paper, the preprocessing of input speech to reduce the bit rate is the different with the conventional vocoder. The different kinds of parameter are used for the preprocessing so this paper is compared with theses parameters for finding the more appropriate parameter for the vocoder. The parameters are used to synthesize the speech not to encode or decode for coding technique so we proposed the simple algorithm not to have the influence on the processing time or the computation time. The parameters in used the preprocessing step are speaking rate, duration and PSOLA technique.

• Proceedings of the IEEK Conference
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• 2000.11d
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• pp.199-202
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• 2000

Speaking Rate has variety depends on the situation and habit of speakers. It has been many studied about speaking rate In speaker recognition. The study of speaking rate in speech recognition is one of considerable matter when It is recognized the speakers and it is measured by many speech data base and complicate estimation for accuracy. In this paper, conventional vocoder process the speech signal when encoding and transmitting without regard to speaking rate so in order to apply the speaking rate for vocoder It should be considered the simpler algorithm and less computation amount than the conventional method of speaking rate used In speech recognition. We proposed the speaking rate algorithm which is used the simple parameter with Line Spectrum Pair (LSP). The proposed peaking rate method is measured by the information of LSP in speech. We measured the variety rate of phenomenon about utterances which have different velocity, respectively. As a result, It has distinct variation rate of phenomenon between utterances uttered fast and slow and the rate is 42.8% higher in case of uttered fast than in case of uttered slow.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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• pp.957-962
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• 1994

Real-time RELP vocoder is implemented on the TMS320C25 DSP chip. The implemented system is IBM-PC add-on board and composed of analog in/out unit, DSP unit, memoy unit, IBM-PC interface unit and its supporting assembly software. Speech analyzer and synthesizer is implimented by DSP assembly software. Speech parameters such as LPC coefficients, base-band residuals, and signal gains is extracted by autocorrelation method and inverse filter and synthesized by spectral folding method and direct form synthesis filter in this board. And then, real-time RELP vocoder with 9.6Kbps is simulated by down-loading method in the DSP program RAM.

• The Journal of the Acoustical Society of Korea
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• v.20 no.3E
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• pp.18-24
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• 2001

Speaking rate represents how many phonemes in speech signal have in limited time. It is various and changeable depending on the speakers and the characters of each phoneme. The preprocessing to remove the effect of variety of speaking rate is necessary before recognizing the speech in the present speech recognition systems. So if it is possible to estimate the speaking rate in advance, the performance of speech recognition can be higher. However, the conventional speech vocoder decides the transmission rate for analyzing the fixed period no regardless of the variety rate of phoneme but if the speaking rate can be estimated in advance, it is very important information of speech to use in speech coding part as well. It increases the quality of sound in vocoder as well as applies the variable transmission rate. In this paper, we propose the method for presenting the speaking rate as parameter in speech vocoder. To estimate the speaking rate, the variety of phoneme is estimated and the Line Spectrum Pairs is used to estimate it. As a result of comparing the speaking rate performance with the proposed algorithm and passivity method worked by eye, error between two methods is 5.38% about fast utterance and 1.78% about slow utterance and the accuracy between two methods is 98% about slow utterance and 94% about fast utterances in 30 dB SNR and 10 dB SNR respectively.