• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.8C
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• pp.575-582
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• 2008

In this paper we described the process and the results of real-time implementation of G.729.1 wideband speech codec which is standardized in SG15 of ITU-T. To apply the codec on ARM926EJ-S(R) processor core. we transformed some parts of the codec C program including basic operations and arithmetic functions into assembly language to operate the codec in real-time. G.729.1 is the standard wideband speech codec of ITU-T having variable bit rates of $8{\sim}32kbps$ and inputs quantized 16 bits PCM signal per sample at the rate of 8kHz or 16kHz sampling. This codec is interoperable with the G.729 and G.729A and the bandwidth extended wideband($50{\sim}7,000Hz$) version of existing narrowband($300{\sim}3,400Hz$) codec to enhance voice quality. The implemented G.729.1 wideband speech codec has the complexity of 31.2 MCPS for encoder and 22.8 MCPS for decoder and the execution time of the codec takes 11.5ms total on the target with 6.75ms and 4.76ms respectively. Also this codec was tested bit by bit exactly against all set of test vectors provided by ITU-T and passed all the test vectors. Besides the codec operated well on the Internet phone in real-time.

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.230-235
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• 2008

This paper addresses issues associated with the real-time implementation of a wideband speech codec such as ITU-T G. 729. 1 on an ARM processor in order to provide an improved voice quality of a VoWiFi service. The real-time implementation features in optimizing the C-source code of G.729. 1 and replacing several parts of the codec algorithm with faster ones. The performance of the implementation is measured by the CPU time spent for G.729.1 on the ARM926EJ processor that is used for a VoWiFi phone. It is shown from the experiments that the G.729.1 codec works in real-time with better voice quality than G 729 codec that is conventionally used for VoIP or VoWiFi phones.

• The Journal of the Acoustical Society of Korea
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• v.28 no.4
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• pp.392-400
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• 2009

In this paper, we propose an effective speech reinforcement technique base on ITU-T G.729A CS-ACELP codec under the near-end background noise environments. In general, since the intelligibility of the far-end speech for the near-end listener is significantly reduced under near-end noise environments, we require a far-end speech reinforcement approach to avoid this phenomena. In contrast to the conventional speech reinforcement algorithm, we reinforce the excitation signal of the codec's parameters received from the far-end speech signal based on the G.729A speech codec under various background noise environments. Specifically, we first estimate the excitation signal of ambient noise at the near-end through the encoder of the G.729A speech codec, reinforcing the excitation signal of the far-end speech transmitted from the far-end. we specially propose a novel approach to directly reinforce the excitation signal of far-end speech signal based on the decoder of the G.729A. The performance of the proposed algorithm is evaluated by the CCR (Comparison Category Rating) test of the method for subjective determination of transmission quality in ITU-T P.800 under various noise environments and shows better performances compared with conventional SNR Recovery methods.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.10B
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• pp.939-945
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• 2008

Tn this paper we described the process and the results of an implementation of Internet terminal using G.729.1 wideband speech codec for next generation network. For this purpose firstly we chose a high performance RISC application processor having DSP features for speech codec processing and enhanced Multimedia Accelerator(eMMA) function for video codec. In the implementation of this terminal, we used G.729.1 codec recently standardized in ITU-T which is a new scalable speech and audio codec that extends 0.729 speech coding standard. To adopt G.729.1 codec to this terminal we transformed most of the fixed point C codes which require more complexity into assembly codes so as to minimize processing time in the processor. As a result of this work we reduced the execution time of the original C codes about 80% and operated in real time on the terminal. For video we used H.263/MPEG-4 codec which is supported by the eMMA with hardware in the processor. In the SIP call processing test connected to real network we obtained under looms end-to-end delay and 3.8 MOS value measured with PESQ instrument. Besides this terminal operated well with commercial terminals.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.16 no.2
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• pp.69-76
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• 2023

A high-quality audio/voice codec using the MDCT/IMDCT process can perfectly restore the current frame through an overlap-add process with the previous frame. In the overlap-add process, an algorithm delay equal to the frame length occurs. In this paper, we propose a MDCT/IMDCT process that reduces algorithm delay by using a variable phase shift in MDCT/IMDCT process. In this paper, a low-delay audio/speech codec was proposed by applying the low delay MDCT/IMDCT algorithm to the ITU-T standard codec G.729.1 codec. The algorithm delay in the MDCT/IMDCT process can be reduced from 20 ms to 1.25 ms. The performance of the decoded output signal of the audio/speech codec to which low-delay MDCT/IMDCT is applied is evaluated through the PESQ test, which is an objective quality test method. Despite of the reduction in transmission delay, it was confirmed that there is no difference in sound quality from the conventional method.

• ETRI Journal
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• v.31 no.4
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• pp.408-418
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• 2009

Adopting an encryption function in voice over Wi-Fi service incurs problems such as additional power consumption and degradation of communication quality. To overcome these problems, a partial encryption (PE) algorithm for compressed speech was recently introduced. However, from the security point of view, the partial encryption sets (PESs) of the conventional PE algorithm still have much room for improvement. This paper proposes a new selection method for finding a smaller PES while maintaining the security level of encrypted speech. The proposed PES selection method employs the perceptual evaluation of the speech quality (PESQ) algorithm to objectively measure the distortion of speech. The proposed method is applied to the ITU-T G.729 speech codec, and content protection capability is verified by a range of tests and a reconstruction attack. The experimental results show that encrypting only 20% of the compressed bitstream is sufficient to effectively hide the entire content of speech.

• 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 the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.687-695
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• 2002

Media-specific FEC techniques, suggested to confront with VoIP speech packet loss, improve speech quality at the expense of generating additional one-frame delay. In this paper, we suggest new media-specific FEC, i.e, LSF FEC technique which is able to improve speech quality with much shortened additional delay. In the proposed technique, the LSF parameters of the future frame are utilized to recover a lost packet. To evaluate performance of the proposed technique, we use ITU-T G.723.1 and G.729 Codec and apply Gilbert packet loss model and estimate MOS per every packet loss rate using PESQ speech quality estimation algorithm. The proposed technique has effect of shortening delay over from 6.5ms to 27ms compared with existing media-specific FEC techniques. Simulation results for comparison of reconstructed speech quality show this novel technique improves the MOS over 0.1 in practical lossy environment of 5 % packet loss rate.

• Journal of Korea Multimedia Society
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• v.7 no.7
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• pp.896-904
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• 2004

In this paper, We present an efficient variable rate speech coder for spectral efficiency and packet processing technique for packet loss compensation of a voice codec with 10msec frame in VoIP service. Through disconnecting the users from the spectral resource during silence interval of about 60% period, a variable rate voice coder based on a voice activity detection(VAD) can increase spectral gain by two times. The performance of the method was analyzed by variation of detected voice activity factor and degraded speech frame ratio under various background noise level, and compared those of G.729B of ITU-T 8kbps standard speech codec. A method to compensate lost packets utilized addition of recovery data to a main stream and error concealment scheme for speech quality enhancement, the performance is verified by reconstructed speech quality. The proposed scheme can achieve spectral gain by two times or enhance speech quality by 3dB through reserved bandwidth of VAD. Therefore, the proposed method can enhance a spectral efficiency or speech quality of VoIP.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12A
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• pp.2027-2035
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• 2001

In this paper, we present an audio stream delivery using the AMR (Adaptive Multi-Rate) coder that was adopted by ETSI and 3GPP as a standard vocoder for next generation IMT-2000 service in which includes combined sender (FEC) and receiver reconstruction technique in the Internet. By use of the media-specific FEC scheme, the possibility to recover lost packets can be much increased due to the addition of repair data to a main data stream, by which the contents of lost packets can be recovered. The AMR codec is based on the code-excited linear predictive (CELP) coding model. So we use a frame erasure concealment for CELP-based coders. The proposed scheme is evaluated with ITU-T G.729 (CS-ACELP) coder and AMR - 12.2 kbit/s through the SNR (Signal to Noise Ratio) and the MOS (Mean Opinion Score) test. The proposed scheme provides 1.1 higher in Mean Opinion Score value and 5.61 dB higher than AMR - 12.2 kbit/s in terms of SNR in 10% packet loss, and maintains the communicab1e quality speech at frame erasure rates lop to 20%.