• Journal of Biomedical Engineering Research
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• v.18 no.4
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• pp.457-466
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• 1997

We developed a new hearing aid telephone which helps the hearing impaired person to improve the listening ability of telephone speech. Recently, the hearing impaired person and the elderly who has hearing loss have been continuously increased and their desire for participating society as a producer has been increased also. So they strong1y want the hearing aid devices which make compensation fortheir handicap. The hearing aid telephone is one of the basic aid devices that helps the hearing impaired to communicate well with other poeple and to acquire easily useful information through the phone. We analyze the hearing ability of the hearing impaired, design the new model of the hearing aid telephone and test the telephone in three fields-electrical, word perception, user test. Our new tolephone has lour band pass filter channels and the center frequencies of these filters are 500, 1000, 2000, 3000Hz which are considered psychoacoustic factors and telephone line characteristics. The hearing impaired can adjust the total gain characteristics of receiving sound to his hearing ability by setting four volumes in the telelphone. This procedure is called fitting which is a very important factor for the hearing impaired to take meaning of speech. The total gain of this telephone is over 20dB from 250Hz to 3200Hz range. From the results of the tests we certify that our new model is better for the hearing impaired to understand the meaning or telephone speech than the old general models. The next step of developing the hearing aid telephone is to study about compressing sidetone and noise, dividing frequency bands, selecting hearing aid pattern and compensating psychoacoustic loudness. we expect that the advanced hearing aid telephone can be developed by the research about speech perception characteristics of the hearing impaired in engineering and clinical side.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.16 no.9 s.114
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• pp.897-903
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• 2006

The reduction of the vehicle interior noise has been the main interest of noise and vibration harshness (NVH) engineers. The driver's perception on the vehicle noise is affected largely by psychoacoustic characteristic of the noise as well as the SPL. In particular, the heating, ventilation and air conditioning (HVAC) system sound among the vehicle interior noise has been reflected sensitively in psychoacoustics view point. Even though the HVAC noise is not louder than overall noise level, it clearly affects subjective perception to drivers in the way of making to be nervous or annoyed. Therefore, these days a vehicle engineer takes aim at developing sound quality as well as reduction of noise. In this paper, we acquired noises in the HVAC from many vehicles. Through the objective and subjective sound quality (SQ) evaluation with acquiring noises recorded by the vehicle HVAC system, the simple and multiple regression models were obtained for the subjective evaluation 'Pleasant' using the semantic differential method (SDM). The regression procedure also allows you to produce diagnostic statistics to evaluate the regression estimates including appropriation and accuracy. Furthermore, the neural network (NN) model were obtained using three inputs(loudness, sharpness and roughness) of the SQ metrics and one output(subjective 'Pleasant'). Because human's perception is very complex and hard to estimate their pattern, we used NN model. The estimated models were compared with correlations between output indexes of SQ and hearing test results for verification data 'Pleasant'. As a result of application of the SQ indexes, the NN model was shown with the largest correlation of SQ indexes and we found possibilities to predict the SQ metrics.

• Journal of Broadcast Engineering
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• v.25 no.2
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• pp.233-241
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• 2020

In this paper, a new buffer control method is proposed for improving the performance of the frequency domain part of the 3D audio (3DA) core codec. For the proposed buffer control method, we first combine the 3DA RM9 with the 3GPP AAC buffer control method which includes the psychoacoustic model and rate-distortion control process with the spectral hole avoidance algorithm. Then, we revise the 3GPP buffer control method so as to achieve a faithful bit allocation to the frames with higher activity. With the MUSHRA test, we prove that the proposed buffer control method demonstrates better performance than the 3DA RM9 and 3GPP AAC.

• Journal of Broadcast Engineering
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• v.6 no.2
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• pp.195-202
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• 2001

As a solution for copyright protection of digital multimedia contents, digital watermark technology is now drawing the attention. In this paper, we presented two novel audio watermarking algorithms as a solution for protecting unauthorized copy of digital audio. Proposed watermarking schemes include the psychoacoustic model of MPEG audio coding to achieve the perceptual transparency after watermark embedding and preprocessing procedure before correlation in watermark detection to extract copyright information without access to the original audio signal. Experimental results show that our watermarking scheme is robust to common signal Processing attacks and it Introduces no audible distortion after watermark insertion.

• The Journal of the Acoustical Society of Korea
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• v.19 no.3
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• pp.11-19
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• 2000

This research deals with the implementation procedures of a chorus processing DSP for karaoke system. It is necessary to compress the chorus data to store as many choruses as we can. We apply MPEG-1 audio algorithm to compress the chorus data. And the chorus system must be accompanied with the karaoke that can change the key and the tempo. So the chorus DSP must be able to change the key and tempo of the chorus data. We apply SOLA (Synchronized Overlap and Add) to do it. We designed the chorus DSP that can compress the chorus, change the key and tempo. And we verified the chorus DSP logic using FPGA. The used FPGA are two FLEX10K100s made by ALTERA. Finally we make the ASIC chip of chorus DSP and verify its operation.

• 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.1
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• pp.62-68
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• 2010

This paper proposes a new masking threshold adjustment strategy to improve the performance for speech signals in low bit-rate audio coding. After determining formant regions, the masking threshold is adjusted by using the energy ratio of each sub-band to the average energy of each formant. More quantization noises are added to the bands that have relatively large energy, but less distortion is allowed in spectral valley regions by allocating more bits, which reflects the concept of perceptual weighting widely used in speech coding. From the results of objective speech quality measure, we verified that the proposed method improves quality for the speech input signals compared to the conventional one.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.1
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• pp.86-96
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• 2013

Unified Speech and Audio Coding (USAC) is the speech/audio codec with the best quality, approved on Final Draft International Standard (FDIS) at MPEG meeting in 2011. Since MPEG conventionally standardizes only the decoder, it is not easy to study on the encoder technologies. Furthermore, Reference Model(RM) shows extremely poor performance. To solve these problems, the open source project(JAME) proposes the methods to make the improved performance of main encoder technologies in USAC. Especially, this paper introduces the encoder modules: the signal classifier for selective operation between two coders, the psychoacoustic model in frequency domain, and window transition technology. Finally, the results of verification test for FDIS and the performance of Common Encoder are appended.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2006.05a
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• pp.1443-1448
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• 2006

The reduction of the vehicle interior noise has been the main interest of NVH engineers. The driver's perception on the vehicle noise is affected largely by psychoacoustic characteristic of the noise as well as the SPL. In particular, the HVAC sound among the vehicle interior noise has been reflected sensitively in the side of psychology. Even though the HVAC noise is not louder than overall noise level, it clearly affects subjective perception in the way of making a diver become nervous or annoyed. Therefore, these days a vehicle engineer takes aim at developing sound quality as well as reduction of noise. In this paper, we acquired noises in the HVAC from many vehicles. Through the objective and subjective sound quality evaluation with acquiring noises caused by the vehicle HVAC system, the simple and multiple regression models were obtained for the subjective evaluation 'Pleasant' using the sound quality metrics. The regression procedure also allows you to produce diagnostic statistics to evaluate the regression estimates including appropriation and accuracy. Furthermore, the neural network model were obtained using three inputs(loudness, sharpness and roughness) of the sound quality metrics and one output(subjective 'Pleasant'). And then the models were compared with correlations between sound quality index outputs and hearing test results for 'Pleasant'. As a result of application of the sound quality index, the neural network was verified with the largest correlation of the sound quality index.

• 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.