• The Journal of the Acoustical Society of Korea
• /
• v.42 no.6
• /
• pp.617-626
• /
• 2023

This study aims to investigate the preferred sound level of water sound for various levels of open-plan-office noise regarding soundscape quality and speech privacy. And assessment of the work efficiency of the water sound. For the laboratory experiment, office noise was recorded using a binaural microphone in a real open-plan office. For the assessment of the soundscape quality and speech privacy, Overall Soundscape Quality (OSQ) and Listening Difficulty (LD) were evaluated under three different sound levels (55 dBA, 60 dBA, and 65 dBA) and five different signal-to-noise ratios (SNR -10 dB, -5 dB, 0 dB, +5 dB, and +10 dB). After the evaluation, the preferred SNR was proposed according to OSQ and LD. For the assessment of to work efficiency of water sound, this study evaluated the cognitive performance of both of the condition noise only and combine the water sound with office noise. The results showed that LD increased as the water sound level increased, but OSQ decreased. When the water sound level was more than the office noise level, the OSQ decreased from noise only. Therefore, considering OSQ and LD, the preferred SNR of water sound was -5 dB for all noise levels. At the preferred level of water sound, the cognitive performance results were shown to decrease at 55 dBA compared to noise only, but at 60 dBA and 65 dBA combine the water sound results were increased than the noise only.

• Journal of Environmental Science International
• /
• v.25 no.9
• /
• pp.1241-1251
• /
• 2016

A minimum threshold for the signal to noise ratio ($SNR_{min}$) has to be set in the data processing system of wind profiler radar (WPR). The data collection rate and the accuracy of the WPR wind vector depend on the $SNR_{min}$. The WPR at Uljin is operated with an $SNR_{min}$ of 1 dB which is a relatively large threshold. We found that the accuracy and the continuity of the WPR wind vector with height were directly related to the variability of the SNR and vertical gradient of the squared refractive index. We investigated a quantitative method for determining a new $SNR_{min}$ for the WPR at Uljin and it was evaluated with radiosonde data. The accuracy and continuity of the wind vector from an SNR of less than 1 dB, began to decrease at an altitude of 3.5 km. Most of the SNR values were less than -3.5 dB in altitudes higher than 3.5 km. We retrieved high-accuracy wind vectors at altitudes over 3 km where measurements were deficient with an $SNR_{min}$ of 1 dB.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
• /
• v.11 no.3
• /
• pp.429-436
• /
• 2000

In this paper, the performance of a orthogonal noncoherent FH-BFSK communication system in the presence of partial band noise jamming is analyzed. Also bit error rate(BER) is studied when jamming bandwidth ratio to overall spread spectrum bandwidth,$\rho$, changes according to processing gain(PG). The performance is investigated by numerical analysis and computer simulation of SPW. Even if PG is high, required performance could not gain because error floor occurs when JSR is 10 dB and SNR is under 10 dB. PG can be obtained to acquire a required BER according to $\rho$ when SNR is above 12 dB.

• Journal of the Korean Institute of Intelligent Systems
• /
• v.15 no.2
• /
• pp.230-235
• /
• 2005

In this paper, we propose an algorithm that improves a tone quality of a noisy audio signal in order to enhance a performance of perceptual filter using intelligent estimation of noise pattern from a band degraded by additive noise. The proposed method doesn't use the estimated noise which is obtained from silent range. Instead new estimated noise according to the power of signal and effect of noise variation is considered for each frame. So the noisy audio signal is enhanced by the method which controls a estimation of noise Pattern effectively in a noise corruption band. To show the performance of the proposed algorithm, various input signals which had a different signal-to-noise ratio(SNR) such as $5\cal{dB},\;10\cal{dB},\;15\cal{dB}\;and\;20\cal{dB}$ were used to test the proposed algorithm. we carry out SSNR and NMR of objective measurement and MOS test of subjective measurement. An approximate improvement of $7.4\cal{dB},\;6.8\cal{dB},\;5.7\cal{dB},\;5.1\cal{dB}$ in SSNR and $15.7\cal{dB},\;15.5\cal{dB},\;15.2\cal{dB},\;14.8\cal{dB}$ in NMR is achieved with the input signals, respectively. And we confirm the enhancement of tone quality in terms of mean opinion score(MOS) test which is result of subjective measurement.

• The Journal of the Acoustical Society of Korea
• /
• v.13 no.1E
• /
• pp.82-88
• /
• 1994

It is important to exactly detect the GCI(Glottal Closure Instant) in the speech signal processing. A few methods to detect the GCI of voiced speech have een proposer, untill now. But these are difficult to detect the GCI for wide range of speakers and or various vowel signals. In this paper, we prposed a new method for GCI detection using the G-peak. The speech waveforms are passed through the LPF of variable bandwidth. Then, the GCI's of voiced speech are detected by the G-peak based on the filtered signals. We compared the detected with the eye-checked GCI at the SNR of clean, 20dB, and 0dB. We took into account the range within 1ms between eye-checked and detected GCI. We obtained the result of the detection rate as 97.9% in the clean speech, 96.5% in 20dB SNR, and 94.8% in 0dB SNR, respectively.

• Journal of Digital Convergence
• /
• v.13 no.10
• /
• pp.313-318
• /
• 2015

A traditional speech enhancement methods distort the sound spectrum generated according to estimation of the remaining noise, or invalid noise is a problem of lowering the speech recognition performance. In this paper, we propose a speech detection method that convergence the sound energy distribution process and sound energy parameters. The proposed method was used to receive properties reduce the influence of noise to maximize voice energy. In addition, the smaller value from the feature parameters of the speech signal The log energy features of the interval having a more of the log energy value relative to the region having a large energy similar to the log energy feature of the size of the voice signal containing the noise which reducing the mismatch of the training and the recognition environment recognition experiments Results confirmed that the improved recognition performance are checked compared to the conventional method. Car noise environment of Pause Hit Rate is in the 0dB and 5dB lower SNR region showed an accuracy of 97.1% and 97.3% in the high SNR region 10dB and 15dB 98.3%, showed an accuracy of 98.6%.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.27 no.4A
• /
• pp.330-335
• /
• 2002

In the direct sequence code division multiple access system, the problem of multiple access interference due to multiple access is generated. A interference cancelling receiver is used to solve this problem. The conventional interference cancelling receiver is structure of successive interference canceller using antenna array. In this structure, the difference of between method I and method II depends on updating weight vector. In this paper, the adaptive blind CMA array interference cancelling receiver using cost function of constant modulus algorithms is proposed to update weight vector at conventional structure. The simulation compared the proposed interference cancelling receiver with two conventional interference cancelling receivers by signal to interference ratio and bit error rate curve under additive white Gaussian noise environment. The simulation results show that the proposed receiver has about the gain of SIR of 1.5[dB] more than method I which is conventional receiver at SIR curve, and about the gain of SIR of 0.5(dB) more than method II. In BER curve, the proposed IC receiver about the gain of SNR of 2[dB] more than method I and about the gain of SNR of 0.5[dB] more than method If, Thus, the proposed interference cancelling receiver has the higher performance than conventional interference cancelling receivers.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.29 no.2C
• /
• pp.213-221
• /
• 2004

A new method for design of two-channel finite-impulse response(FIR) quadrature mirror-image filter(QMF) banks with low reconstruction delay using weighting function is proposed. The weighting function used in this paper is calculated from the previous updated filter coefficients vector which is adjusted from iteration to iteration in the design of QMF banks. In this paper, passband and stopband edge frequency are used in design of QMF banks with low delay characteristic in time domain instead of specific frequency interval where the artifacts occur in conventional design method. The investigation of specific frequency interval where artifacts occur can not be required by using passband and stopband edge frequency. Some comparisons of performance are made with other existing design method to demonstrate the proposed method for QMF bank design. and it was observed that the proposed method using the weighted function and passband and stopband edge frequency improves the peak reconstruction error by 0.001 [dB], the peak-to-peak passband ripple by 0.003[dB], SNR with a white noise by 7[dB] and SNR with a step input by 32[dB], but with a reduction of the computational efficiency because of updating the weighting function over the conventional method in Ref [11].

• Journal of the Korea Society of Computer and Information
• /
• v.15 no.2
• /
• pp.109-117
• /
• 2010

Digital watermarking has drawn extensive attention for protecting digital contents from unauthorized copying. This paper proposes a new watermarking scheme in frequency domain for copyright protection of digital audio. In our proposed watermarking system, the original audio is segmented into non-overlapping frames. Watermarks are then embedded into the selected prominent peaks in the magnitude spectrum of each frame. Watermarks are extracted by performing the inverse operation of watermark embedding process. Simulation results indicate that the proposed scheme is robust against various kinds of attacks such as noise addition, cropping, resampling, re-quantization, MP3 compression, and low pass filtering. Our proposed watermarking system outperforms Cox's method in terms of imperceptibility, while keeping comparable robustness with the Cox's method. Our proposed system achieves SNR (signal-to-noise ratio) values ranging from 20 dB to 28 dB. This is in contrast to Cox's method which achieves SNR values ranging from only 14 dB to 23 dB.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.35 no.11C
• /
• pp.911-916
• /
• 2010

This paper proposes a new algorithm to cancel the acoustic feedback and noise signals in digital hearing aids. The proposed algorithm combines the feedback canceller to remove acoustic feedback signals and the noise canceller to reduce background noises. The feedback canceller is implemented by normal adaptive FIR filter, and the noise canceller is implemented by using the Wiener solution in frequency domain. This noise canceller has the transfer function presented by the power spectral density of signals. To verify the performances of the proposed algorithm, the simulations were carried out for the system. As the results of simulations, it was proved that we can advance 10.85dB output SNR on the average for the forward path gain of 0dB, and 11.04dB output SNR on the average for the forward path gain of 6dB, in the case of using the proposed algorithm.