• 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.

• MALSORI
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• no.52
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• pp.121-132
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• 2004

This paper is aiming at real-time implementation of adaptive noise cancellation system using DSP processor. ACHARF algorithm, which guarantees stability and fast convergence by adaptive compensator, is used on this DSP system. For the experiments, TLV320AIC23 stereo CODEC of TI Inc. is used with TMS320C6413 DSP processor. Signals of primary input and reference input are obtained by two microphones. The primary input is the voice plus noise signal and the reference input is white noise or real noise. The experimental results show that ANC system using DSP processor with ACHARF is verified to be an effective speech enhancement method for various speech processing units.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.98-101
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• 2000

On CELP type Vocoders G.723.1 6.3kbps/5.3kbps Dual Rate Speech Codec, which is developed for Internet Phone and videoconferencing, uses VAD(Voice Activity Detection)/CNG (Comfort Noise Generator) in order to reduce the bit rate in a silence period. In order to reduce the bit rate effectively in this paper, we first set the boundary condition of the energy threshold to prevent the consumption of unnecessary processing time, and use three decision rules to detect an active frame by energy, pitch gain and LSP distance. To evaluate the performance of the proposed algorithm we use silence-inserted speech data with 0, 5, 10, 20dB of SNR. As a result when SNR is over 5dB, the bit rate is reduced up to about 40% without speech degradation and the processing time is additionally decreased.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.142-147
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• 1998

Recently, 8 kb/s CS-ACELP coder of G.729 is atandardized by ITU-T SG15 and it has been reported that the speech quality of G729 is better than or equal to that of 32kb/s ADPCM. However G.729 is the fixed rate speech coder, and it does not consider the property of voice activity in mutual conversation. If we use the voice activity, we can reduce the average bit rate in half without any degradations of the speech quality. In this paper, we propose an efficient variable rate algorithm for G.729. The variable rate algorithm consists of two main subjects, the rate determination algorithm and algorithm, we combine the energy-thresholding method, the phonetic segmentation method by integration of various feature parameters obtained through the analysis procedure, and the variable hangover period method. Through the analysis of noise features, the 1 kb/s sub rate coder is designed for coding the background noise signal. So, we design the 4 kb/s sub rate coder for the unvoiced parts. The performance of the variable rate algorithm is evaluated by the comparison of speed quality and average bit rate with G.729. Subjective quality test is also done by MOS test. Conclusively, it is verified that the proposed variable rate CS-ACELP coder produced the same speech quality as G.729, at the average bit rate of 4.4 kb/s.

• Journal of Broadcast Engineering
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• v.10 no.3
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• pp.261-269
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• 2005

Throughout the history of digital communication, the digital speech coder is used as speech compression tool. Nowadays, the speech coder has been rapidly developed in the area of mobile communication system to overcome severe channel error and limitation of radio frequency resources. Due to the development of high performance communication system, high quality of speech coder is needed. This kind of speech coder can be used not only in communication services but also in digital multimedia services. In this paper, we describe the technologies of digital speech coder which are used in wire and wireless communication. We also present a summary of recent speech coding standards for narrowband and wideband applications. Finally we introduce the technical trends of next generation speech coder.

• The Journal of the Acoustical Society of Korea
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• v.29 no.3E
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• pp.131-139
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• 2010

This paper proposes a novel spectral hole substitution technique for low bit-rate audio coding. The spectral holes frequently occurring in relatively weak energy bands due to zero bit quantization result in severe quality degradation, especially for harmonic signals such as speech vowels. The enhanced aacPlus (EAAC) audio codec artificially adjusts the minimum signal-to-mask ratio (SMR) to reduce the number of spectral holes, but it still produces noisy sound. The proposed method selectively predicts the spectral shapes of hole bands using either intra-band correlation, i.e. harmonically related coefficients nearby or inter-band correlation, i.e. previous frames. For the bands that have low prediction gain, only the energy term is quantized and spectral shapes are replaced by pseudo random values in the decoding stage. To minimize perceptual distortion caused by spectral mismatching, the criterion of the just noticeable level difference (JNLD) and spectral similarity between original and predicted shapes are adopted for quantizing the energy term. Simulation results show that the proposed method implemented into the EAAC baseline coder significantly improves speech quality at low bit-rates while keeping equivalent quality for mixed and music contents.

• The Journal of the Acoustical Society of Korea
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• v.28 no.5
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• pp.471-476
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• 2009

In this paper, we apply a discriminative weight training to a support vector machine (SVM) based speech/music classification for the selectable mode vocoder (SMV) of 3GPP2. In our approach, the speech/music decision rule is expressed as the SVM discriminant function by incorporating optimally weighted features of the SMV based on a minimum classification error (MCE) method which is different from the previous work in that different weights are assigned to each the feature of SMV. The performance of the proposed approach is evaluated under various conditions and yields better results compared with the conventional scheme in the SVM.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2018.10a
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• pp.490-492
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• 2018

This paper presents an analysis of the AMR speech data as a basic work to study the technique of inputting and transmitting AMR voice data which is widely used in the public cell phone. AMR consists of HEADER and Speech Data, and it is transmitted in bit format and has 8 bit-rate modes in total. HEADER contains mode information of Speech Data, and the length of Speech Data differs depending on the mode. We chose the best mode which is best to input into QR code and analyzed that mode. It is a goal to show a higher compression ratio for voice data by the analysis and experiments. This analysis shows improvement in that it can transmit voice data more effectively.

• ETRI Journal
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• v.35 no.5
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• pp.935-938
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• 2013

This letter presents an algorithm for selecting a low delay for the modified discrete cosine transform (MDCT) and inverse MDCT (IMDCT). The implementation of conventional MDCT and IMDCT requires a 50% overlap-add (OLA) for a perfect reconstruction. In the OLA process, an algorithmic delay in the frame length is employed. A reduced overlap window and MDCT/IMDCT phase shifting is used to reduce the algorithmic delay. The performance of the proposed algorithm is evaluated by applying the low-delay MDCT to the G.729.1 speech codec.

• Electronics and Telecommunications Trends
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• v.10 no.4 s.38
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• pp.129-142
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• 1995

본 고에서는, 음성을 디지털로 부호화하여 전송함으로써 발생되는 신호 대 양자화왜곡 비(Q)의 개념 및 CODEC과의 관계를 분석하고, MNRU를 디지털 회로로 구현하는데 필요한 입력음성 신호레벨, 잡음의 통계적 성질 및 진폭제한이 음성품질에 미치는 영향을 살펴보았다. 또한, 본 연구에서 구현한 MNRU의 성능에 대해 주관평가 실험을 실시하여, 다른 나라의 주관평가 결과와 비교/분석하였다.