• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.2
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• pp.192-201
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• 1993

According to technical advances of telecommunication, communication network has changed to digital transmission from analog transmission network. In the long run, current network will be altered into ISDN which makes end-to-end digital communication. This transition of communication network brings about an important questions for networking plan, administration, and speech quality in order to achieve the effective and advanced telecommunication. Speech quality criterions and degradation factors of digital communication system differ from those of existing analog system because of other characteristics like single echo. It is, therefore, necessary to design new criterions and specifications for digital communication network. This Paper describes the relation between speech communication and speech transmission quality and describes the implementation of model system for quality evaluation of digital speech communication network. In addition, some applications of model system implemented are proposed.

• MALSORI
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• no.61
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• pp.75-86
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• 2007

In this paper, we propose a scalable audio coder, which has a variable bandwidth from the narrowband speech bandwidth to the audio bandwidth and also has a bit-rate from 8 to 320 kbits/s, in order to cope with the quality of service(QoS) according to the network load. First of all, the proposed scalable coder splits bandwidth of the input audio into narrowband up to around 4 kHz and above. Next, the narrowband signals are compressed by a speech coding method compatible to an existing standard speech coder such as G.729, and the other signals whose bandwidth is above the narrowband are compressed on the basis of a psychoacoustic model. It is shown from the objective quality tests using the signal-to-noise ratio(SNR) and the perceptual evaluation of audio quality(PEAQ) that the proposed scalable audio coder provides a comparable quality to the MPEG-1 Layer III (MP3) audio coder.

• The Journal of the Acoustical Society of Korea
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• v.29 no.5
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• pp.347-353
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• 2010

In VoIP system, the speech quality is seriously degraded due to packet loss, and the degree of degradation by each packet loss depends on the characteristics of the corresponding packet. Therefore, it is possible to improve the speech quality of VoIP by selectively controlling the packet to be lost during transmission based on the expected degradation by the loss of each packet. In this paper, a new scheme to improve speech quality of DiffServ-based VoIP by assigning priority to each packet is proposed, and a method to determine the priority of each packet is developed. The performance of proposed method was measured in packet loss environment based on Gilbert model, and it was verified both objectively and subjectively that the speech quality is improved by the proposed method.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.21 no.12
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• pp.2291-2297
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• 2017

This paper describe to extract speech measure algorithm for evaluating a speech database, and presents generating method of a speech quality measure using DNN(Deep Neural Network). In our previous study, to produce an effective speech quality measure, we propose a combination of various speech measures which are highly correlated with WER(Word Error Rate). The new combination of various types of speech quality measures in this study is more effective to predict the speech recognition performance compared to each speech measure alone. In this paper, we describe the method of extracting measure using DNN, and we change one of the combined measure from GMM(Gaussican Mixture Model) score used in the previous study to DNN score. The combination with DNN score shows a higher correlation with WER compared to the combination with GMM score.

• Phonetics and Speech Sciences
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• v.13 no.3
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• pp.71-78
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• 2021

Most neural-network-based speech synthesis models utilize neural vocoders to convert mel-scaled spectrograms into high-quality, human-like voices. However, neural vocoders combined with mel-scaled spectrogram prediction models demand considerable computer memory and time during the training phase and are subject to slow inference speeds in an environment where GPU is not used. This problem does not arise in linear spectrogram prediction models, as they do not use neural vocoders, but these models suffer from low voice quality. As a solution, this paper proposes a Tacotron 2 and Transformer-based linear spectrogram prediction model that produces high-quality speech and does not use neural vocoders. Experiments suggest that this model can serve as the foundation of a high-quality text-to-speech model with fast inference speed.

• Proceedings of the KIEE Conference
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• 1998.07g
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• pp.2480-2482
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• 1998

In this paper, the implementation of dual rate ADPCM using G.723 16Kbps and 40Kbps speech codec algorithm is handled. For small signals, the low rate 16Kbps coding algorithm shows the same SNR as the high rate 40Kbps coding algorithm, while the low rate 16Kbps coding algorithm shows the lower SNR than the high rate 40Kbps coding algorithm for large signal. To obtain the good trade-off between the data rate and synthesized speech quality, we applied low rate 16Kbps for the small signal and high rate 40Kbps for the large signal. Various threshold values determining the rate are tested for good trade off data rate and speech quality. Also the low pass filter effect of speech input and output devices is simulated at several cut-off frequencies. To simulation result shows the good speech quality at a low rate comparing with 16Kbps & 40Kbps.

• MALSORI
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• v.68
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• pp.49-63
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• 2008

This paper describes the development and assessment of a hidden Markov model (HMM) based Tagalog speech synthesis system, where Tagalog is the most widely spoken indigenous language of the Philippines. Several aspects of the design process are discussed here. In order to build the synthesizer a speech database is recorded and phonetically segmented. The constructed speech corpus contains approximately 89 minutes of Tagalog speech organized in 596 spoken utterances. Furthermore, contextual information is determined. The quality of the synthesized speech is assessed by subjective tests employing 25 native Tagalog speakers as respondents. Experimental results show that the new system is able to obtain a 3.29 MOS which indicates that the developed system is able to produce highly intelligible neutral Tagalog speech with stable quality even when a small amount of speech data is used for HMM training.

• Proceedings of the KSPS conference
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• 2006.05a
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• pp.131-134
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• 2006

The narrowband speech over the telephone network is lacking in the information from low-band (0-300 Hz) and high-band (3400-8000 Hz) that are found in wideband speech (0-8000 Hz). As a result, narrowband speech is characterized by the reduced intelligibility and muffled quality, and degraded speaker identification. Spectral folding is the easiest way to reconstruct the missing high-band; however, the reconstructed speech still brings the sense of band-limited characteristic because of the absence of low-band and mid-band frequency components. To compensate for the lack of the extended speech, we propose to combine the spectral folding method and GMM transformation method, which is a statistical method to reconstruct wideband speech. The reconstructed wideband speech showed that the absent frequency components was filled up with relatively low spectral mismatch. According to the subjective speech quality evaluations, the proposed method was preferred to other methods.

• Phonetics and Speech Sciences
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• v.15 no.4
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• pp.45-51
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• 2023

Voice issues such as voice weakness that arise with age can have social and emotional impacts, potentially leading to feelings of isolation and depression. This study aimed to investigate the changes in acoustic characteristics resulting from aging, focusing on voice quality and spoken fluency. To this end, tasks involving sustained vowel phonation and paragraph reading were recorded for 20 elderly and 20 young participants. Voice-quality-related variables, including F0, jitter, shimmer, and Cepstral Peak Prominence (CPP) values, were analyzed along with speech-fluency-related variables, such as average syllable duration (ASD), articulation rate (AR), and speech rate (SR). The results showed that in voice quality-related measurements, F0 was higher for the elderly and voice quality was diminished, as indicated by increased jitter, shimmer, and lower CPP levels. Speech fluency analysis also demonstrated that the elderly spoke more slowly, as indicated by all ASD, AR, and SR measurements. Correlation analysis between voice quality and speech fluency showed a significant relationship between shimmer and CPP values and between ASD and SR values. This suggests that changes in spoken fluency can be identified early by measuring the variations in voice quality. This study further highlights the reciprocal relationship between voice quality and spoken fluency, emphasizing that deterioration in one can affect the other.

• Phonetics and Speech Sciences
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• v.10 no.4
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• pp.163-171
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• 2018

A mixed dysarthria with combinations of hypokinetic, ataxic, and spastic components is a common clinical feature of multiple system atrophy (MSA). Due to the rapid progress of dysarthria after diagnosis, people with MSA experience difficulty with verbal communication, which eventually affects their quality of life negatively. In this study, SPEAK $OUT!^{(R)}$, an intensive 1:1 treatment of dysarthria for improving functional communicative ability, was provided to twelve people with MSA. To evaluate the efficacy of SPEAK $OUT!^{(R)}$ in people with MSA, aerodynamic, acoustic, and perceptual analyses were conducted. Pre-and post-therapy data included maximum phonation time, vocal intensity, and fundamental frequency during /a/ sustained phonation and passage reading; frequency range between high /a/ and low /a/ phonation; jitter, shimmer, and HNR for vocal quality; speech rate during passage reading; and perceptual evaluation scores for articulation precision and intonation. The participants achieved statistically significant improvement in vocal intensity, pitch range, vocal quality, speech rate, and speech intelligibility. In conclusion, SPEAK $OUT!^{(R)}$ is a feasible treatment for people with MSA to efficaciously improve their speech ability.