• Journal of the Institute of Convergence Signal Processing
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• v.2 no.4
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• pp.53-58
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• 2001

In this paper to develop objective speech quality measure for CDMA telephone network environments, recent developed measures are investigated first. But those measures show low performances in CDMA telephone networks. To solve this problem, new objective speech quality measure adopting noise masking threshold is proposed and studied. To acquire better performance, scaled noise masking threshold calculation for speech signals is employed instead of conventional tone signals. To verify effectiveness of proposed method performance comparison experiments are carried out for CDMA telephone network speech databases, for the results proposed methods show improved performances compared to existing meaures.

• MALSORI
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• no.49
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• pp.63-75
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• 2004

Most existing telephone speech transmitted in current public networks is band-limited to 0.3-3.4 kHz. Compared with wideband speech(0-8 kHz), the narrowband speech lacks low-band (0-0.3 kHz) and high-band(3.4-8 kHz) components of sound. As a result, the speech is characterized by the reduced intelligibility and a muffled quality, and degraded speaker identification. Bandwidth extension is a technique to provide wideband speech quality, which means reconstruction of low-band and high-band components without any additional transmitted information. Our new approach considers to exploit harmonic synthesis method for reconstruction of low-band speech over the CELP coded speech. A spectral distortion measurement and listening test are introduced to assess the proposed method, and the improvement of synthesized speech quality was verified.

• 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.

• Proceedings of the IEEK Conference
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• 2007.07a
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• pp.423-424
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• 2007

In this paper, we propose a scalable high-quality speech reconstruction method for distributed speech recognition (DSR). It is difficult to reconstruct speech of high quality with MFCCs at the DSR server. Depending on the bit-rate available by the DSR system, we can send additional information associated with speech coding to the DSR sorrel, where the bit-rate is variable from 4.8 kbit/s to 11.4 kbit/s. The experimental results show that the speech quality reproduced by the proposed method when the bit-rate is 11.4 kbit/s is comparable with that of ITU-T G.729 under both ideal channel and frame error channel conditions while the performance of DSR is maintained to that of wireline speech recognition.

• The Journal of the Acoustical Society of Korea
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• v.18 no.7
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• pp.10-16
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• 1999

In this paper, we propose a concatenation-based speech synthesizer using CVC(consonant-vowel-consonant) speech segments extracted from an undesigned continuous speech corpus. Natural synthetic speech can be generated by a proper modelling of coarticulation effects between phonemes and the use of natural prosodic variations. In general, CVC synthesis unit shows smaller acoustic degradation of speech quality since concatenation points are located in the consonant region and it can properly model the coarticulation of vowels that are effected by surrounding consonants. In this paper, we analyze the characteristics and the number of required synthesis units of 4 types of speech synthesis methods that use CVC synthesis units. Furthermore, we compare the speech quality of the 4 types and propose a new synthesis method based on the most promising type in terms of speech quality and implementability. Then we implement the method using the speech corpus and synthesize various examples. The CVC speech segments that are not in the speech corpus are substituted by demonstrate speech segments. Experiments demonstrate that CVC speech segments extracted from about 100 Mbytes continuous speech corpus can produce high quality synthetic speech.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.30B no.10
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• pp.42-49
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• 1993

In this paper, we evaluate the quality of synthesized speech by rule using the LPC CD as a objective measure, and then compare the test result with the subjective one. Speech used for the test consists of 108 words which are selected by word construction method using Korean attribute and frequency distribution, synthesized by demi-syllable rule. By evaluating the quality of synthesized speech by reule objectively, we have tried to resolve the problems such as lots of evaluation time, expansion of test scale, and variables of analysis result arised by subjective measure. We have, also, proved the validity of the objective test using the LPC CD, by comparing intelligibility which is the index for the subjective quality evaluation of synthesized speech by rule with MOS. From this results, we can provide a guide for quality assessment that would be useful in the R&D of synthesis method and the commercial products using synthesized speech.

• Proceedings of the KIEE Conference
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• 2000.07d
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• pp.3166-3168
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• 2000

This paper proposed variable threshold dual rate ADPCM coding method which is modified from the standard ADPCM of ITU G.726 for speech quality improvement. The speech quality of variable threshold dual rate ADPCM is better than single rate ADPCM at noisy environment without increasing the complexity by using ZCR(Zero Crossing Rate). In this case, ZCR is used to divide input signal samples into two categories(noisy & speech). The samples with higher ZCR is categorized as the noisy region and the samples with lower ZCR is categorized as the speech region. Noisy region uses higher threshold value to be compressed by 16Kbps for reduced bit rates and the speech region uses lower threshold value to be compressed by 40Kbps for improved speech quality. Comparing with the conventional ADPCM, which adapts the fixed coding rate. the proposed variable threshold dual rate ADPCM coding method improves noise character without increasing the bit rate. For real time applications, ZCR calculation was considered as a simple method to obtain the background noise information for preprocess of speech analysis such as FFT and the experiment showed that the simple calculation of ZCR can be used without complexity increase. Dual rate ADPCM can decrease the amount of transferred data efficiently without increasing complexity nor reducing speech quality. Therefore result of this paper can be applied for real-time speech application such as the internet phone or VoIP.

• Journal of Korea Multimedia Society
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• v.5 no.4
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• pp.405-414
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• 2002

In this paper, we propose a new objective speech quality measure using noise masking threshold for speech quality assessment of mobile telephone network environments, and verify the effectiveness of the proposed method through the experiments. For such a purpose, well known objective speech quality measures such as BSD and PSQM are first evaluated for digital mobile telephone network environments. However, these conventional methods does not have good performance under mobile networks environments compared to literary results. To be mote effective objective speech quality measure under mobile telephone environments, the proposed method employs human psychoacoustic masking effect. The DMOS, instead of MOS, is used as a subjective speech quality measure for performance evaluation. The performance comparison are carried out with speech data collected from digital mobile telephone environments. As results, the proposed measure have and average 4% higher performance, in terms of correlation, than existing objective speech quality measures such as BSD and PSQM.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.2
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• pp.279-287
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• 2001

In this paper, we discuss the scale of subjective speech quality measurement over VoIP(Voice over IP) network which is a component of broadband networks. Objective parameters of multimedia services like PSNR or jitter can easily measured and defined, but these factors are not easily meet the user's perceptual recognition. We suggest the speech quality measurement scale through the subjective measurement for end-to-end speech quality composed of sender-side quality, transmission quality, receiver-side quality, which provide the degree of correctness of representation of speaker, the degree of impairment caused by various factors, the degree of recognition of processed speech, respectively. Also, we examined the proposed method and verify it's availability.

• Phonetics and Speech Sciences
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• v.4 no.4
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• pp.87-92
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• 2012

Speech enhancement aims to improve speech quality by removing background noise from noisy speech. Independent vector analysis is a type of frequency-domain independent component analysis method that is known to be free from the frequency bin permutation problem in the process of blind source separation from multi-channel inputs. This paper proposed a new method of microphone array based speech enhancement that combines independent vector analysis and beamforming techniques. Independent vector analysis is used to separate speech and noise components from multi-channel noisy speech, and delay-sum beamforming is used to determine the enhanced speech among the separated signals. To verify the effectiveness of the proposed method, experiments for computer simulated multi-channel noisy speech with various signal-to-noise ratios were carried out, and both PESQ and output signal-to-noise ratio were obtained as objective speech quality measures. Experimental results have shown that the proposed method is superior to the conventional microphone array based noise removal approach like GSC beamforming in the speech enhancement.