• Journal of Industrial Convergence
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• v.21 no.8
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• pp.23-33
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• 2023

Consumer review analysis is important for product development, customer satisfaction, competitive advantage, and effective marketing. Increased use of wireless earphones is expected to reach $45.7 billion by 2026 with growth in lifestyle. Therefore, in consideration of the growth and importance of the market, consumer reviews of wireless earphones from Apple and Samsung were analyzed. In this study, 11,320 wireless earphone reviews from Apple and Samsung sold on Coupang were collected to analyze consumers' purchase intentions and analyze consumer satisfaction through analysis of the frequency, sensitivity, and LDA topic model of text mining. As a result of topic modeling, 16 topics were derived and classified into sound quality, connection, shopping mall service, purchase intention, battery, delivery, and price. As a result of brand comparison, Samsung purchased a lot for gift purposes, had a high positive sentiment for price, and Apple had a high positive sentiment for battery, sound quality, connection, service, and delivery. The results of this study can be used as data for related industries as a result of research that can obtain improvements and insights on customer satisfaction, quality and market trends, including manufacturing, retail, marketers, and consumers.

• The Journal of the Acoustical Society of Korea
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• v.26 no.4
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• pp.159-165
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• 2007

In this paper, we propose a new pre-selection of candidate units that is suitable for the unit selection based Japanese TTS system. General pre-selection method performed by calculating a context-dependent cost within IP (Intonation Phrase). Different from other languages, however. Japanese has an accent represented as the height of a relative pitch, and several words form a single accentual phrase. Also. the prosody in Japanese changes in accentual phrase units. By reflecting such prosodic change in pre-selection. the qualify of synthesized speech can be improved. Furthermore, by calculating a context-dependent cost within accentual phrase, synthesis speed can be improved than calculating within intonation phrase. The proposed method defines AP. analyzes AP in context and performs pre-selection using accentual phrase matching which calculates CCL (connected context length) of the Phoneme's candidates that should be synthesized in each accentual phrase. The baseline system used in the proposed method is VoiceText, which is a synthesizer of Voiceware. Evaluations were made on perceptual error (intonation error, concatenation mismatch error) and synthesis time. Experimental result showed that the proposed method improved the qualify of synthesized speech. as well as shortened the synthesis time.

• The Journal of the Acoustical Society of Korea
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• v.31 no.7
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• pp.419-426
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• 2012

The pitch searching is a vital process in a vocoder. Generally, the method of pitch searching is employed after highlighting the periodicity, where a correlation is identified with the signal by changing the interval of two pulses. When the correlation value reaches the peak, the pitch can be found by the pulse interval because it is the repetition interval with most striking period. However if the identified period happens to be one of half period, double period or triple period, this cannot be considered as the pitch period. Many methods were suggested to solve this problem. An inaccurate pitch could be obtained as well, when there is an interval where signal amplitude is not constant but varies abruptly in the frame. To solve this matter, searching the pitch by dividing a frame into various subframes is adopted, but too much calculation has to be followed while it leads the correct value. This paper suggests an algorithm to resolve these two problems. First, to search the pitch after advance correction of the signal energy level with an estimated overall energy change ratio in the frame before pitch search to reduce half period, double period and triple period is suggested. Second, to vary the number of subframes by predicting the amplitude change rate in the frame by the energy ratio obtained by the above-mentioned method is advised. If these two methods are applied, the pitch searching time can be reduced and the general pitch searching performance can be improved without affecting the sound quality in the synthesized signal.

• The Journal of the Korea institute of electronic communication sciences
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• v.8 no.12
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• pp.1793-1802
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• 2013

Various approaches for sound source segregation have been proposed. Among these approaches, frequency domain binaural model(FDBM) has the advantages of low computational load and effective howling cancellation. A binaural hearing assistance system based on FDBM has been proposed. This system can enhance desired signal based on the directivity information. Although FDBM has been evaluated in terms of signal-to-noise ratio (SNR) and coherence function, the evaluation results do not always agree with the human impressions. These evaluation methods provide physical measures, and do not take account of perceptual aspect of human being. Considering a binaural hearing assistance system as a one of major applications, the quality of segregated sound should keep level enough. In the paper, signal segregation performance by means of FDBM is evaluated by three objective methods, i.e., SNR, coherence and Perceptual Evaluation of Speech Quality(PESQ), to discuss the characteristic of FDBM on the sound source segregation performance. The simulation's evaluation results show that FDBM improves the quality of the left and right channel signals to an equivalent level. And the results suggest the possibility that PESQ provides a more useful measure than SNR and coherence in terms of the segregation performance of FDBM. The evaluation results by PESQ show the effects from segregation parameters and indicate appropriate parameters under the conditions. In the paper, signal segregation performance by means of FDBM is evaluated by three objective methods, i.e., SNR, coherence and PESQ, to discuss the characteristic of FDBM on the sound source segregation performance. The simulation's evaluation results show that FDBM improves the quality of the left and right channel signals to an equivalent level. And the results suggest the possibility that PESQ provides a more useful measure than SNR and coherence in terms of the segregation performance of FDBM. The evaluation results by PESQ show the effects from segregation parameters and indicate appropriate parameters under the conditions.

• The Journal of the Acoustical Society of Korea
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• v.32 no.4
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• pp.346-353
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• 2013

This paper deals with the implementation of a high-quality audio system for karaoke. For improving the key/tempo changes performance, we separated the audio into many musical instrument channels. By separating musical instrument channels, high-quality key/tempo changes can be achieved and we confirmed this using the cross-correlation distribution and the MOS evaluation. The improved audio system was implemented using the TMS320C6747 DSP with fixed/floating-point operations. The implemented audio system can perform the multi-channel WMA decoding, the MP3 encoding/decoding, the wav playing, the EQ, and the key/tempo changes in real time. The WMA channels used for processing the separated instrument channels. The audio system includs the MP3 encoding/decoding function for playing and recording and the wav channel for the effect sound.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.10 no.1
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• pp.102-110
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• 2017

In this paper, we have implemented an real-time adaptive feedback cancellation(AFC) algorithm that can be applied to multi-channel digital hearing aid. Multichannel digital hearing aid typically use the FFT filterbank based Wide Dynamic Range Compression(WDRC) algorithm to compensate for hearing loss. The implemented real-time acoustic feedback cancellation algorithm has one integrated structure using the same FFT filter bank with WDRC, which can be beneficial in terms of computation affecting the hearing aid battery life. In addition, when the AFC fails to operate due to nonlinear input and output, the reduction gain is applied to improve robustness in practical environment. The implemented algorithm can be further improved by adding various signal processing algorithm such as speech enhancement.

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.218-223
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• 2008

In this paper, we implement several methods to improve an audio collaboration system over IP networks, and then evaluate the performance of the implemented methods. In general, speech and audio quality degrades depending on the characteristics of IP networks such as jitter and packet loss. In order to reduce this quality degradation, we propose a lower bit rate audio delivery scheme using the MPEG-2 AAC (Advanced Audio Coding) audio codec in a viewpoint that a packet loss rate could be reduced by a smaller packet size. In addition, iLBC (Internet Low-Bitrate Codec) and the G.711 packet loss concealment algorithm defined by IEFT and ITU-T, respectively, are applied to a audio collaboration system. RAT (Robust-Audio Tool)[7] is used as a baseline platform for the implementation of the proposed methods. It is shown from the implementation that the implemented MPEG-2 AAC audio codec with a bitrate of 256 kbit/s performs as similar as the uncompressed audio quality with a bitrate of 512 kbit/s, and that iLBC and the G.711 packet loss concealment algorithm can improve speech quality when a packet loss rate is 2~10%.

• Journal of Broadcast Engineering
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• v.11 no.2 s.31
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• pp.191-199
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• 2006

This paper presents an efficient pre echo control scheme for MPEG Audio coders based on the psychoacoustic model II (PAM-II). Pre echo control is the final step for the calculation of masking threshold in the PAM II. It is to minimize the spread of quantization error over the processing frame. In the conventional encoders, pre echo is reduced by restricting the estimated masking threshold not to exceed the one obtained in the previous frame. The conventional method performs pre echo control not only for short blocks but also for long blocks, which lowers the masking threshold in long blocks and, in turn, increases the quantization noise level of corresponding blocks. This paper proposes an efficient pre echo control process. The test result shows a mean enhancement of more than 0.4 especially for complex signals on the ITU R 5 point audio impairment scale.

• The Journal of the Acoustical Society of Korea
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• v.15 no.6
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• pp.110-116
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• 1996

In this paper a novel method improving the accuracy of DTMF tone reception in ARS system is proposed. ARS system should allow users to generate DTMF signals while it is sending voice guidance. It is not unocmmon, in this case, that a portion of transmitting voice signals cross-talks to the receiving channel and it often results in interfering with the receiving DTMF signals. Serious degradations including DTMF tone missing, false alarm and so forth have been introduced for the above reason. To overcome this phenomena, we have proposed a way eliminating the frequency spectra representing DTMF signals bands from the transmitting voice signal by using notch filters. This method also employs bandpass filters of which the frequency responses are reciprocal to those of the notch filters incorporated with the DTMF receiver. It is shown that a drastic improvement has been achieved with respect to the DTMF tone detection with little deterioration of voice guidance quality.

• The Journal of the Acoustical Society of Korea
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• v.31 no.1
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• pp.29-38
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• 2012

This paper presents a transient noise reduction system in a speech signal. The proposed transient noise reduction system utilizes a median filter to reduce the transient noise. Since the median filter can distort speech during the noise reduction, a long-term prediction (LTP) filter is adopted as a pre-processor to minimize speech distortion. The speech information preserved by the LTP filter is re-synthesized after reducing the noise. This paper verifies the weakness of a linear prediction (LP) filter and the superiority of the LTP filter for preserving the speech component in transient noise presence environment. Applying the proposed system, the signal-to-noise ratio (SNR) of output is improved by 8dB in both speech and noise presence region, and PESQ score is increased by 1 point comparing with noisy input.