• 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 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.

• The Journal of the Acoustical Society of Korea
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• v.20 no.7
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• pp.63-68
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• 2001

This paper describes the details of ITU-T SGIS G.729A speech coder implementation using ARM9 Thumb/sup R/ processor core and various techniques used in the optimization process. ITU-T G.729 speech coder is the standard of the toll quality 8 kbit/s speech coding. The input to the speech encoder is assumed to be a 16 bits PCM signal at a sampling rate of 8000 samples per second. G.729A is reduced complexity version of the G.729 coder. This version is bit stream interoperable with the full version. The implemented coder requires 34.8 MIPS for the encoder and 8.1 MIPS for the decoder, 36.5 kBytes of program ROM and 6.3 kBytes of data RAM, respectively. The implemented coder is tested against the set of 9 test vectors provided by ITU-T for bit exact implementation.

• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.939-942
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• 2001

본 논문에서는 ITU-T G.723.1, G.729 부호화기와 MLT(Modulated Lapped Transform) 방법을 이용한 광대역 음성 부호화방법을 제안한다. 제안된 광대역 음성부호화 방법은 16 kHz로 샘플링된 입력신호를 QMF(Quadrature Mirror Filter)사용하여 저대역과 고대역으로 나누며, 각 대역은 8 kHz의 샘플링을 갖는 협대역 음성 신호로 변환된다. 고대역은 MLT변환 후 벡터 양자화하며 또한 MLT를 사용한 ATC(Adaptive Transform Coding)방법을 적용하여 표현하며 저대역은 G.723.1과 G.729 부호화기를 사용한다. 설계된 광대역 음성부호화기의 성능을 평가하기 위하여 MOS (Mean Opinion score)실험을 수행하였다. MOS 실험을 통해 16 kbps G.729-MLT VQ방식이 G.722 56kbps 와 비슷한 음질을 나타내었다.

• 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 the Acoustical Society of Korea
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• v.19 no.4
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• pp.45-51
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• 2000

This paper describes real-time implementation of a multi-channel G.729A speech coder using a 16 bit fixed-point Digital Signal Processor (DSP) and its application to a Voice Mailing Service (VMS) system. TMS320C549 by Texas Instruments was used as a fixed point DSP chip and a 4 channel G.729A coder was implemented on the chip. The implemented coder required 14.5 MIPS for the encoder and 3.6 MIPS for the decoder at each channel. In addition, memories required by the coder were 9.88K words and 1.69K words for code and data sections, respectively. As a result, the developed VMS system that accommodates two DSP chips was able to support totally 8 channels. Experimental results showed that the our multi-channel coder passes all of test vectors provided by ITU-T.

• Phonetics and Speech Sciences
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• v.1 no.4
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• pp.97-103
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• 2009

This paper describes a new algorithm for encoding spectral envelope in the time domain alias cancellation (TDAC) part of G.729.1. The spectral envelope and modified discrete cosine transform (MDCT) coefficients of the weighted code-excited linear predictive (CELP) coding error in lower-band and the higher-band input signal are encoded in the TDAC part. In order to reduce allocation bits for spectral envelope coding, a new algorithm using sub-band correlation between adjacent frames is proposed. In addition, to improve the quality of decoded signals, two bit allocation strategies using reduced bits from the proposed algorithm are proposed. The performance of the proposed algorithm is evaluated in terms of objective quality and bit reduction rates. Experimental results show that the proposed algorithm increases the quality of sounds significantly.

• 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.

• Electronics and Telecommunications Trends
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• v.21 no.6 s.102
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• pp.77-85
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• 2006

2006년 5월 고품질 인터넷 전화(VoIP) 서비스에 사용할 수 있는 가변대역 광대역 음성 코덱 기술이 ITU-T SG16 회의에서 국제 표준으로 확정되었다. ITU-T에서 표준화된 G.729.1 코덱은 국내 IPR이 포함된 최초 음성 코덱 분야의 표준이다. 이 코덱은 인터넷 전화 서비스에서 가장 널리 사용되는 G.729를 기반으로 8-32kbps 범위 내에서 임베디드 형태로 비트열 계층을 쌓아가는 구조로 협대역 신호(300-3400Hz)부터 광대역 신호(50-7000Hz)를 압축, 복원한다. 이 기술은 기존의 인터넷 전화에서 사용하고 있는 코덱의 낮은 품질, 대역폭 확장, 품질 제어가 곤란한 단점을 개선하여 인터넷전화 서비스 시장을 활성화 할 것으로 기대된다.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.45 no.2
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• pp.81-89
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• 2008

Wideband speech, characterized by a bandwidth of about 7 kHz (50-7000 Hz), provides a substantial quality improvement in terms of naturalness and intelligibility. Although higher data rates are required, it has extended its application to audio and video conferencing, high-quality multimedia communications in mobile links or packet-switched transmissions, and digital AM broadcasting. In this paper, we present a new bandwidth-scalable coder for wideband speech and audio signals. The proposed coder spits 8kHz signal bandwidth into two narrow bands, and different coding schemes are applied to each band. The lower-band signal is coded using the ITU-T G.729/G.729E coder, and the higher-band signal is compressed using a new algorithm based on the gammatone filter bank with an invertible auditory model. Due to the split-band architecture and completely independent coding schemes for each band, the output speech of the decoder can be selected to be a narrowband or wideband according to the channel condition. Subjective tests showed that, for wideband speech and audio signals, the proposed coder at 14.2/18 kbit/s produces superior quality to ITU-T 24 kbit/s G.722.1 with the shorter algorithmic delay.