• The Journal of the Acoustical Society of Korea
• /
• v.32 no.2
• /
• pp.157-166
• /
• 2013

This paper proposes a non-intrusive speech quality estimation method considering the effects of packet loss to perceptual quality. Packet loss is a major reason of quality degradation in a packet based speech communications network, whose effects are different according to the input speech characteristics or the performance of the embedded packet loss concealment (PLC) algorithm. For the quality estimation system that involves packet loss effects, we first observe the packet loss of G.729 codec which is one of narrowband codec in VoIP system. In order to quantify the lost packet affects, we design a classification algorithm only using speech parameters of G.729 decoder. Then, the degradation values of each class are iteratively selected that maximizes the correlation with the degradation PESQ-LQ scores, and total quality degradation is modeled by the weighted sum. From analyzing the correlation measures, we obtained correlation values of 0.8950 for the intrusive model and 0.8911 for the non-intrusive method.

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

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

• Proceedings of the IEEK Conference
• /
• 2003.07e
• /
• pp.2248-2251
• /
• 2003

Communication between speech networks employing different speech codecs requires interoperability. The cascade connection of two different codecs, called tandem coding, not only degrades speech quality, but also produces high computational loads. These Problems can be solved by using the transcoding algorithm. This paper presents an effective algorithm for transcoding from G.729A to EVRC and its simulation results.

• Proceedings of the KSPS conference
• /
• 2007.05a
• /
• pp.307-310
• /
• 2007

In this paper, a packet loss concealment (PLC) algorithm for CELP-type speech coders is proposed to improve the quality of decoded speech under a burst packet loss condition. The proposed algorithm is based on the recovery of voiced excitation using an estimate of the voicing probability and the generation of random excitation by permutating the previously decoded excitation. The voicing probability is estimated from the correlation using the previous correctly decoded excitation and pitch. The proposed algorithm is implemented as a PLC algorithm for G.729 and its performance is compared with PLC employed in G.729 by means of perceptual evaluation of speech quality (PESQ) and an A-B preference test under the random and burst packet losses with rates of 3% and 5%. It is shown that the proposed algorithm provides better speech quality than the PLC of G.729, especially under burst pack losses.

• Proceedings of the IEEK Conference
• /
• 2000.09a
• /
• pp.791-794
• /
• 2000

본 논문에서는 TMS320C549 를 사용하여 4 채널 G.729A 음성 부호화기를 실시간 구현하였으며, 실제로 음성 사서함 서비스 시스템에 응용하였다. 구현된 G.729A 는 패널 당 부호화기와 복호화기에 각각 14.5MIPS 와 3.6 MIPS 를 소요하였으며, 메모리는 코드와 데이터 부분에 각각 9.88K 워드, 1.69 K 워드를 필요로 하였다. 결과적으로 개발된 VMS 시스템에는 두 개의 DSP 를 사용하여 DSP 당 4 채널씩 총 8 채널을 수용하였다. 실험 결과, ITU-T에서 제공된 모든 테스트 벡터 결과와 비트 단위로 동일하였다.

• The Journal of the Acoustical Society of Korea
• /
• v.21 no.3
• /
• pp.293-303
• /
• 2002

This paper implements optimized Time Scale Modification (TSM) system using ITU G.729 speech decoder and Dual SOLA algorithm. The proposed system assume 8 Kz sampling rate, 80 samples/frame input speech from the ITU G.729 speech Decoder and the TSM (Time Scale Modification) feature of Dual SOLA produces the high quality output speech that was slow-down or speed up as a user's choice. Especially, the proposed Optimized Dual SOLA base on various simulations and theoretical analysis, and the additional interpolation procedure of the speech makes it possible to setup high performance integrated TSM system at the maximum time scale modification rate. The system performance is analyzed and verified with various input speech and playback speed.

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

• The Journal of the Korea institute of electronic communication sciences
• /
• v.7 no.2
• /
• pp.243-248
• /
• 2012

In this paper we simulate the capacity of VOIP calls through WiFi network by computer simulations using OPNET modeler. The results show that sudden quality degradations occur on all VoIP calls when the number of call of an AP(Access Point) increases beyond a specific value. The reason of the quality degradation was turned out to be the queueing delay at the down link of AP. Under the IEEE 802.11b environments, the maximum number of VoIP calls of an AP maintaining the required voice quality (MOS > 2.5), was evaluated as 5, 12, and 27 when we use G.711, G.729a, and G.729a VAD codec, respectively.

• Proceedings of the IEEK Conference
• /
• 2000.09a
• /
• pp.823-826
• /
• 2000

유/무선 통신 시스템에서는 통신망마다 각각 다른 음성 부호화기를 사용하므로 음성신호는 두 번의 부/복호화 과정을 거치게 된다. 이로 인해 음질저하, 계산량 증가, 그리고 전달 지연 증가 등의 문제가 발생된다. 본 논문에서는 위의 문제점들을 개선하기 위하여 패킷 음성통신과 무선 이동 통신에 사용되는 음성 부호화기의 상호부호화를 위한 알고리듬을 제안한다 효율적인 음성 패킷 변환 방법을 제안하였으며, 8 kbps G.729A 패킷을 5.3 kbps G.723.1 패킷으로 변환하는 방법을 제안한다. 제안된 음성 패킷 변환 방법은 LSP 변환과정, 적응코드북 변환과정 그리고 고정 코드북 고속 탐색 과정으로 구성된다. 여러 가지 음성 신호로 모의 실험한 결과, 본 논문에서 제안된 상호부호화 알고리듬이 두 번의 부/복호화 과정을 거친 경우보다 짧은 전달 지연 시간과 적은 계산량으로 동등한 음질의 음성신호로 복호화함을 확인하였다.