• 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.

• Journal of KSNVE
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• v.9 no.5
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• pp.991-997
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• 1999

It is getting higher the demand on a good sound in a car. Like some other sensory qualities, achieving a good sound is still treated as a black-box followed by the individual taste of tuning engineers. This study aims to define the attributes of a good car audio sound and the assessment factors affecting the sound quality. With a variety of surveys on professional listening groups and car audio installers, 26 most frequently used vocabularies are selected as typical presentations of car audio sound quality. After repetitions of listening tests with deliberately selected test CD, pre-recorded 15 different digital car-recordings, headphones replay system, and three different listening groups, it is proved that there are at least 5 different assessment factors which significantly affect the car audio sound quality. They are high fidelity, spectral balance, tonal accuracy, sound stage and acoustic fault.

• Proceedings of the KSPS conference
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• 2006.11a
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• pp.131-134
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• 2006

Quality of narrowband speech $(0{\sim}4kHz)$ can be enhanced by the bandwidth expansion technique, by which the high- band components are estimated. This paper proposes the bandwidth expansion method using the spline codebook based spectral folding. For the performance evaluation, the PESQ(Perceptual Evaluation of Speech Quality) scores are measured as the objective measurement In addition, the MOS (Mean Opinion Score) and the preference tests are performed as the subjective measurement. The results show our proposed method outperforms the existing spline based one.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.4
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• pp.156-163
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• 1996

In this paper, we described the implementation and the sound quality assessment of a real-time stereo audio codec using TMS320C40 DSP (digital signal processing) chip for low bitrte and high quality audio. We implemented hardware and software in order to overcome a real-time processing problem of audio compression algorithm that can be produced by largely recursive computing and complexity of the process. We have studied five types of distortion that can be produced by perceptual coding and the codec was evaluated by eight test musics that are selected in SQAM (sound quality assessment material) 422-2-4-2 produced by EBU (european broadcast union). The subjective listening tests were carried out on the codec quality and preformance by double blind method in a listening room with eleven listeners. As a result, 5 grade-impairment scale was scored under minus one and the codec quality was evaluated to be perceptible, but not annoying.

• Proceedings of the KSPS conference
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• 2007.05a
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• pp.315-318
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• 2007

This paper proposes the methods to enhance the speech quality of source controlled variable bit-rate coder based on the waveform interpolation. The methods are to estimate and generate the parameters that are not transmitted from encoder to decoder by the repetition and extrapolation schemes. For the performance evaluation, the PESQ(Perceptual Evaluation of Speech Quality) scores are measured. The experimental results shows that our proposed method outperforms the conventional source controlled variable bit-rate coder. Especially, the performance of the extrapolation method is better than that of the repetition method.

• The Journal of the Acoustical Society of Korea
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• v.18 no.8
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• pp.108-115
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• 1999

In this study we suggested a digital filter method to enhance car audio sound quality against the sound distortion due to cabin's acoustic characteristics and car driving noises. The digital filters designed were based on the characteristics on car driving noises and cabin acoustic characteristics. Car driving noises were analyzed by two ways; one is an objective method, octave band frequency analysis method. The other is a subjective method; sensory evaluation method, NCB method. On these results, seven sets of modified coefficients of eleven band digital filters were obtained. To find optimum audio sound quality among nine sound samples filtered by designing seven types of digital filters, which were mixed car driving noises at 100km/h, subjective evaluation method was used, paired comparison method; Scheffe' seven point method.

• The Journal of the Acoustical Society of Korea
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• v.15 no.2
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• pp.55-60
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• 1996

This paper describes the subjective assessment of the multi-channel audio quality according to the bitrate changing and evaluates the predistortion effect to avoid the unmasked noise after matrixing/dematrxing process in transmission and regeneration of the multi-channel audio. The simulation is processed by the perceptual coding that is MPEG-2 Audio layer II algorithm. We evaluate the quality improvement about predistortion using or not by 384, 320, 256, 128kbps. As the result of the double blind subjective assessment, 5 Grade-Impairment Scale is scored under minus one to 320kbps and so audio quality is evaluated to be perceptible, but not annoying in 3/2 channel. The effect of the predistortion is improved one level in 128kbps and especially speech test material I better improved than music test materials.

• Proceedings of the KSPS conference
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• 2004.05a
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• pp.235-238
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• 2004

Frame erasures cause speech quality degradation in wireless communication networks or packet networks. The degradation becomes worse when consecutive frame erasures occur. Speech coders have a frame erasure concealment(FEC) mechanism to compensate for frame erasures. It is meaningful to evaluate the performance of FEC mechanisms for frame erasures that occur in communications networks. In this paper, various frame erasures are designed. And the FEC algorithms of speech coders are evaluated and analyzed with the Perceptual Evaluation of Speech Quality(PESQ). It is found that the performances vary in accordance with frame erasure types, frame erasure rates, and utterance lengths.

• The KIPS Transactions:PartC
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• v.11C no.6 s.95
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• pp.773-780
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• 2004

In this paper, we measured the transmission characteristics of a MPEG2 and a DV that are typical stream-type traffics on the very high speed network and carried out the subjective evaluation of end users for these stream-types. In the subjective evaluation of these stream-type data, video quality evaluation is based on ITU-R BT.500-1 and audio qualify evaluation is based on ITU-R BS.1116-1. Also experiment method to acquire the subjective evaluation of end users is selected the 5 grades method of DSCQS. Under the same condition, in case of MPEG2, the evaluation grade of the video and the audio duality becomes deteriorated at the load rate of $54\%$ that network traffic increases rapidly. In case of DV the evaluation grade of video duality began decrease, but the degree of the change was slower than MPEG2 at the same load rate. Moreover the subjective evaluation grade of end users was superior to load rate $70\%$ in case of DV audio quality, traffic and QoS control that consider the subjective evaluation of end user is required. Conclusively, in case of MPEG2, we can perform traffic control that only use the actual measurement values on the network. However in case of DV, we can perform traffic control that the actual measurement values on the network and the subjective evaluation of end users are considered at the same time.

• The Journal of the Acoustical Society of Korea
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• v.26 no.3
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• pp.109-114
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• 2007

In this paper, we Propose a pyramid VQ to quantize the transform coefficients of TCX module for the audio improvement of AMR-WB+ codec. The Proposed pyramid VQ is compared to the $RE_8$ Lattice VQ used in the AMR-WB+ standard codec. demonstrating improvement 4% and 5.7%. respectively, in Mean Squared Error (MSE) and 3.3% and 4.7%. respectively, in Perceptual Evaluation of Audio Quality (PEAQ) by 8-dimensional and 16-dimensional Pyramid VQ.