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Thorell, Hampus. "Voice Activity Detection in the Tiger Platform." Thesis, Linköping University, Department of Electrical Engineering, 2006. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-6586.
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<p>Sectra Communications AB has developed a terminal for encrypted communication called the Tiger platform. During voice communication delays have sometimes been experienced resulting in conversational complications.</p><p>A solution to this problem, as was proposed by Sectra, would be to introduce voice activity detection, which means a separation of speech parts and non-speech parts of the input signal, to the Tiger platform. By only transferring the speech parts to the receiver, the bandwidth needed should be dramatically decreased. A lower bandwidth needed implies that the delays slowly should disappear. The problem is then to come up with a method that manages to distinguish the speech parts from the input signal. Fortunately a lot of theory on the subject has been done and numerous voice activity methods exist today.</p><p>In this thesis the theory of voice activity detection has been studied. A review of voice activity detectors that exist on the market today followed by an evaluation of some of these was performed in order to select a suitable candidate for the Tiger platform. This evaluation would later become the foundation for the selection of a voice activity detector for implementation.</p><p>Finally, the implementation of the chosen voice activity detector, including a comfort noise generator, was done on the platform. This implementation was based on the special requirements of the platform. Tests of the implementation in office environments show that possible delays are steadily being reduced during periods of speech inactivity, while the active speech quality is preserved.</p>
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Carbonneau, Patrice. "Transmission vocale via Internet utilisant les codeurs G.729 et G.729a." Mémoire, Sherbrooke : Université de Sherbrooke, 2002. http://savoirs.usherbrooke.ca/handle/11143/1183.
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Leite, Cavalcante Dirceu. "Uma análise comparativa dos codificadores/ decodificadores de voz para comunicações digitais." Universidade Federal de Pernambuco, 2009. https://repositorio.ufpe.br/handle/123456789/5370.
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Previous issue date: 2009<br>O presente trabalho apresenta uma análise comparativa dos codificadores utilizados em comunicações digitais, notadamente em chamadas VoIP, para fonemas da Língua Portuguesa. O foco deste trabalho é a análise da capacidade do processamento se adaptar às variações do trato vocal durante a pronúncia de intervalos contendo fonemas em frases, simulando uma resposta em tempo real durante uma conversação. Para tal, foram extraídas as freqüências fundamental e das três primeiras formantes para cada um dos intervalos para um grupo de homens e mulheres de várias faixas etárias. A criação de tais intervalos e extração das freqüências foram efetuadas através do programa Praat, com a utilização de análise perceptual e espectrogramas. Resultados mostram uma sutil diferença no processamento da freqüência das formantes entre homens e mulheres. Observou-se também um fenômeno de correção da freqüência fundamental em intervalos contendo variantes consonantais tanto para ambos os sexos. Foram analisados os codificadores de voz G.722, G.723.1, G.726, G.728, G.729A, iLBC e Speex
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Trilling, Romain. "Codage large bande de la parole par encapsulation du codeur ITU G.729 (CS-ACELP)." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1998. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape11/PQDD_0002/MQ40628.pdf.
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Khadra, Ali. "Amélioration de la robustesse de décodeurs de parole basés sur le modèle CELP en utilisant les informations retardées : application au standard G.729 pour la voix sur IP = Improving the robustness of CELP-like speech decoders using late-arrival packets information : application to G.729 standard in VoIP." Sherbrooke : Université de Sherbrooke, 2004.
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Khadra, Ali. "Amélioration de la robustesse de décodeurs de parole basés sur le modèle CELP en utilisant les informations retardées application au standard G.729 pour la voix sur IP." Mémoire, Université de Sherbrooke, 2003. http://savoirs.usherbrooke.ca/handle/11143/1229.
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Voice over Internet applications is the new trend in telecommunications and networking industry today. Packetizing data/voice is done using the Internet protocol (IP). Various codecs exist to convert the raw voice data into packets. The coded and packetized speech is transmitted over the Internet. At the receiving end some packets are either lost, damaged or arrive late. This is due to constraints such as network delay (fitter), network congestion and network errors. These constraints degrade the quality of speech. Since voice transmission is in real-time, the receiver can not request the retransmission of lost or damaged packets as this will cause more delay. Instead, concealment methods are applied either at the transmitter side (coder-based) or at the receiver side (decoder-based) to replace these lost or late-arrival packets. This work attempts to implement a novel method for improving the recovery time of concealed speech The method has already been integrated in a wideband speech coder (AMR-WB) and significantly improved the quality of speech in the presence of jitter in the arrival time of speech frames at the decoder. In this work, the same method will be integrated in a narrowband speech coder (ITU-T G.729) that is widely used in VoIP applications. The ITUT G.729 coder defines the standards for coding and decoding speech at 8 kb/s using Conjugate Structure Algebraic Code-Excited Linear Prediction (CS-CELP) Algorithm.
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Kovařík, Jiří. "Detektor řečové aktivity v signálovém procesoru." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2012. http://www.nusl.cz/ntk/nusl-219784.
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In this diploma thesis were created voice activity detectors according to the standard ITU-T G.729 and G.723.1. The voice activity detectors were implements in the digital signal processor TMS320C6416 made by Texas Instruments. At the same time detectors were designed using by MATLAB programming language. The diploma thesis can be divided into two parts. In the theoretical section provides information on how to report detectors in the standard ITU-T G.729 and G.723.1. In the implementation part is described steps in the implementation of the detector in signal processor TMS320C6416 and there are discussed various differences compared to the documentation.
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Adamec, Michal. "Moderní rozpoznávače řečové aktivity." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2008. http://www.nusl.cz/ntk/nusl-217322.
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This masters theses deals with standard detection methods of speech/pause - voice activity detectors are based on the principles of short-time energy, real spectrum, short-time intensity and on a combinations of these three detectors. In the next parts, there are mentioned other voice activity detectors based on hidden Markovov‘s models and a detector described in the ITU-T G.729 standard. All the detectors, mentioned above, were implemented in research environment MATLAB. Further there was created an user interface for testing functions of the implemented detectors. Finally, there was done an evaluation by ROC characteristics according to the results of the testing.
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Lúdik, Michal. "Porovnání hlasových a audio kodeků." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2012. http://www.nusl.cz/ntk/nusl-219793.
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This thesis deals with description of human hearing, audio and speech codecs, description of objective measure of quality and practical comparison of codecs. Chapter about audio codecs consists of description of lossless codec FLAC and lossy codecs MP3 and Ogg Vorbis. In chapter about speech codecs is description of linear predictive coding and G.729 and OPUS codecs. Evaluation of quality consists of description of segmental signal-to- noise ratio and perceptual evaluation of quality – WSS and PESQ. Last chapter deals with description od practical part of this thesis, that is comparison of memory and time consumption of audio codecs and perceptual evaluation of speech codecs quality.
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陳建吉. "Computational Improvement for G.729 Standard." Thesis, 2002. http://ndltd.ncl.edu.tw/handle/79305205596461365434.
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碩士<br>國立臺北科技大學<br>電機工程系碩士班<br>91<br>The computational improvement on G.729 standard is studied in this thesis. Three algorithms are employed and proposed to enhance the computational performance of G.729. Firstly, the “TIE+PDE” algorithm is employed to reduce the VQ-based quantization of LSP parameter. More than 13.97% of computation is reduced. Secondly, the “WD-LSP” algorithm is proposed to reduce the computation of open-loop pitch. More than 23.69% of computation is reduced. Finally, the “NCPL” algorithm is proposed to reduce the search space of fixed codebook. More than 18.38% of computation is reduced. Finally, three algorithms are simultaneously applied, which result in a total reduction of 56.04% in computation, with a reasonable good quality of synthesized speech sound.
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Chen, Guan-Ming, and 陳冠銘. "Search Algorithms for Codebooks of G.729E Speech Coder." Thesis, 2009. http://ndltd.ncl.edu.tw/handle/11237649972139748127.
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碩士<br>南台科技大學<br>資訊工程系<br>97<br>The algebraic code excited linear prediction (ACELP) algorithm due to low complexity and high quality in its analysis-by-synthesis optimization has been adopted by many speech coding standards. To improve the quality of the coded speech, ITU-T G.729E that is based on G.729 coding structure is recommended. This thesis studied the search algorithms of the adaptive-codebook and the stochastic-codebook of ACELP speech coder. In order to reduce the computational complexity of ACELP coders, the focused search, the depth-first tree search and the global pulse-replacement search are suggested in the ITU-T G.729, G.729A and G.729.1, respectively. Based on the global pulse-replacement (GPR) search, this thesis proposed several derivative GPR algorithms to reduce the computational complexity of stochastic-codebook. In addition, this thesis proposed a complexity scalability design for adaptive codebook search. Experimental results show that the proposed approaches can reduce the computational complexity of codebook for the speech coder G.729E with perceptually negligible degradation of the speech quality. The experimental results in the MOS-LQO score of the proposed approaches have reflected the facts.
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Ju, Fu-Shing, and 朱復興. "G.723.1 and G.729 Speech Transmission over Wireless and Internet Environments." Thesis, 2000. http://ndltd.ncl.edu.tw/handle/59828560619539774553.
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碩士<br>國立中央大學<br>電機工程研究所<br>88<br>G.723.1 and G.729 are both ITU-T speech coding standards. These two speech coders are designed for the compression of speech and audio signals in multimedia applications. The target applications are the voice transmission over Internet and the voice coding of H.323 based Internet video conferences.
To enhance the error robustness of G.729, we propose an error protection scheme by applying BCH(31, 26) to protect significant bits in a frame. Furthermore, the BCH(31, 26) scheme is compatible to G.729 standard. In Internet environment, the packets often get lost and the error propagation effect degrades the speech quality substantially. Therefore, we apply the error concealment techniques to both G.729 and G.723.1. In simulations, the objective and subjective evaluation tests all show that the G.729 BCH(31, 26) error protect scheme and the error concealment techniques of G.729 and G.723.1 can improve speech quality significantly.
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Zhuo, Chang Zhi, and 卓長志. "Computational Reduction For G.729’s LSP Quantization." Thesis, 2010. http://ndltd.ncl.edu.tw/handle/31560353809992208867.
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碩士<br>國立勤益科技大學<br>電機工程系<br>98<br>G.729 standard has been widely used in the VoIP system. But the computational complexity is too large to real-time work via software implementation on embedded devices. Even though the G.729A, it is still hard to implement. This paper will focus on the computational reduction of the quantization procedure of LSP coefficients. Thus, the hybrid two-stage VQ is proposed to replace the original structure. Experimental results confirm that more than 80% of computations can be eliminated in comparison with the conventional G.729 as well as the performance of speech quality is still good. The proposed approach can be successfully used in LSP quantization procedure and the efficiency is excellent.
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Huang, Lin-Chi, and 黃麟祺. "Implementation of G.729 Speech Codeon SoC Platform." Thesis, 2006. http://ndltd.ncl.edu.tw/handle/89703324939129316604.
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碩士<br>國立成功大學<br>電腦與通信工程研究所<br>94<br>In this thesis, the implementation of G.729 codec on ARM at SoC platform is our main research. Since the G.729 codec needs a great deal of calculation, the source code have to be optimized and rewritten in assembly code. Then, fast algorithm that accelerates the fixed-codebook search is applied. Finally, we complete a speech storage by using G.729 speech codec working on ARM.
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Huang, Li-Fang, and 黃麗芳. "AMR to G.729A speech transcoding with fast codebook search." Thesis, 2007. http://ndltd.ncl.edu.tw/handle/09292227811549153386.
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碩士<br>國立中央大學<br>通訊工程研究所<br>95<br>As the development of the internet technique, we not only can transmit the data but also connect 3GPP with VoIP over internet . Because of the coding schemes of 3GPP are not the same as VoIP, speech transcoding scheme is needed in the voice system over internet. Speech transcoding scheme can make the connection between users successful, and furthermore, it can be used in entertainment applications, such as audio chat rooms and online games.
Full decoding technique is an intuitive and traditional speech transcoding method, but it requires high computational complexity and long processing time. In this work, we propose a partial decoding technique with fast codebook search, which utilizes the pulse replacement method, on ACELP coding architecture. There is no need to redo all the decoding and encoding processes. Partial decoding method can be directly applied to ACELP based speech coding, such as AMR and G.729A speech standards. It achieves excellent voice quality as the full decoding method does while it only requires 7.2% computation loading on clockticks per frame.
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Lin, Yu-Pin, and 林裕斌. "The Fast Algorithm for ITU-T G.729 and G.723.1 speech coder." Thesis, 2002. http://ndltd.ncl.edu.tw/handle/jwjtf2.
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碩士<br>國立成功大學<br>電機工程學系碩博士班<br>90<br>The main research of the thesis is focused on the ITU-T G.729 and G.723.1 speech coder, which are low bit-rate speech coders used in digital networks. We propose some fast methods to reduce the complexity of G.729 and G.723.1 stochastic codebook searches. The MTP-ACELP and CMP-MLQ are suggested to improve the speed of ACELP and MP-MLQ search mechanisms in the G.723.1 coder. In the G.729 coder, the cross-correlation function d is used to reduce the search loops. Furthermore, we also propose some methods to reduce the complexity in computation of auto-correlation functionΦ.
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Su, Ying-Jang, and 蘇盈彰. "An Efficient Algebraic Codebook Search For G.729 Standard." Thesis, 2009. http://ndltd.ncl.edu.tw/handle/36331534793527535039.
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碩士<br>國立勤益科技大學<br>電子工程系<br>97<br>G.729 standard has been widely used in the VoIP system. But the computational complexity is too large to real-time work via low-end DSP chip or soft-DSP on embedded devices. Even though the G.729A, it is still hard to implement. Therefore, the goal of this paper is to reduce the computational complexity of G.729 encoding phase. In the G.729 encoding phase, the algebraic codebook search is employed to obtain the four pulse positions in the codebook. However, it occupies the largest computation in the encoding phase. A sub-optimal search algorithm is employed here to reduce the computation required for the search procedure of fixed codebook. Two experiments including the computational complexity and the speech with PESQ-MOS score were used to examine the proposed method. In comparison with the conventional algorithm, the 80.283% of computation can be eliminated with a slight degradation of PESQ-MOS from 3.923 to 3.591. Experimental results confirm that the proposed method has been implemented successfully.
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Chang, Jung-Hsien, and 張榮憲. "Design and Implementation of Multirate G.729 Speech Vocoders." Thesis, 2001. http://ndltd.ncl.edu.tw/handle/90679975706139301117.
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碩士<br>國立成功大學<br>電機工程學系<br>89<br>The main idea of this research paper is base on ITU - T G 729 to design multirate speech vocoders, in addition, to focus on simplifying computation of ACELP. In the research, first, to design "G.729D-" and "G.729E+". Therein, "G.729D-" is calculated by LPC SD of neighbor subframes to reduce the bitrate of ITU-T G.729D vocoder. On the other hand, "G.729E+" is built an intensive layer on ITU-T G.729E vocoder in order to improve the quality of the compressed speech. At the end, preselecting and setting up table to reduce complex computing of ACELP.
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Brito, Carla Sofia Inácio. "Implementação do Codificador de Fala G.729 em FPGA." Dissertação, 2014. https://repositorio-aberto.up.pt/handle/10216/72746.
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Brito, Carla Sofia Inácio. "Implementação do Codificador de Fala G.729 em FPGA." Master's thesis, 2014. https://repositorio-aberto.up.pt/handle/10216/72746.
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Yan, Yu-Liang, and 顏毓良. "Fast Implementation of G.723.1 & G.729 Codecs in Single C6x DSP Chip." Thesis, 2000. http://ndltd.ncl.edu.tw/handle/27998847670658865241.
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碩士<br>國立成功大學<br>電機工程學系<br>88<br>Because of fast development of network technology, low-bit-rate and high-quality digital speech communication is widely noticed by general public. The speech related products in communication or entertainment regions are offered on software or hardware format and deeply popular with people. The G.723.1 and G.729 are two such famous speech compression codecs that they are widely employed in many applications. In this thesis, we will introduce these two codecs, and compare the differences of the structure and applications of them, which help us to understand them better. Besides, we speed up the coding and decoding of the G.723.1 codec by using C6x DSP chip to improve it’s efficiency. Finally, since many speech compression codecs use ACELP search method to find the aperiodic part of speech signal, we replace original ACELP search by proposing a fast algorithm to reduce the complexity and to fasten the program.
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Yeong-Ching, Wang, and 王永清. "A Study on the Implementation of G.729 Speech Codec." Thesis, 1998. http://ndltd.ncl.edu.tw/handle/79424077369056409760.
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碩士<br>國立臺灣科技大學<br>電子工程技術研究所<br>86<br>In these years, the idea of the Digital Speech Codec is high speech quality, low bit rate, low coding delay, high robustness and low complexity. However, for current technology, it''s hard to totally meet the above mentioned requirements. In current standardized speech codec, they are developed for same specific applications. ITU have proposed a number of speech codecs for telecommunication applications, such as G.721, G.723, G.728. In this thesis, we studied and implemented the G.729 8kb/s speech codec which was proposed by ITU in 1995. Major topic in the study and implementation are: (1) To verify the accuracy of open-loop pitch period estimation;(2) To compare the time consumption percentage in the Adaptive codebook and the fixed codebook search; (3) To confirm the algebraic codebook advantages in storage and search complexity; (4) To compare the complexity between G.729 and G.729A; (5) To increase one codebook track to improve the speech quality with slightly larger bit rate.
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Zhong, Yue-Huan, and 鍾岳桓. "An Efficient Algebraic Codebook Search for G.729 Speech Codec." Thesis, 2014. http://ndltd.ncl.edu.tw/handle/46867347632342473863.
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碩士<br>國立勤益科技大學<br>電機工程系<br>102<br>In a bid to enhance the search performance, this paper presents an improved version of reduced candidate mechanism (RCM), an algebraic codebook search conducted on an algebraic code-excited linear-prediction (ACELP) speech coder. This improvement is made based on two findings in a piece of our prior work. The first finding is that there is a 0.8321 probability that the number 1 ranked pulse in a global sorting by pulse contribution is indeed one of the optimal pulses, and the second is that the speech quality can be well maintained at an accuracy rate above 50% approximately. Hence the number 1 pulse in the global sorting is labeled as one of optimal pulse, following which a sequence of search tasks are fulfilled through RCM. This proposed complexity reduction algorithm, implemented on a G.729A speech codec, takes as few as 8 searches, a search load tantamount to 2.5% of G.729A, 12.5% of global pulse replacement method (iteration=2), 16.7% of iteration-free pulse replacement method and 50% of RCM (N=2). This proposal is thus found to successfully reduce the required computational complexity to a great extent as intended.
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Yu, Hsin-Min, and 尤新閔. "Data Hiding Techniques for G.729 and MELP Speech Coding." Thesis, 2002. http://ndltd.ncl.edu.tw/handle/68238879291662619967.
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碩士<br>國立中央大學<br>電機工程研究所<br>90<br>Data hiding is the art of hiding secret messages within a multimedia signal. Most data hiding techniques developed today for speech cannot defense the attack of linear predictive coding (LPC), which is widely used in speech communication systems, and that means it is very difficult to transmit the secret messages and speech simultaneously. A different approach is to hide the secret message in the compressed bit stream.
In this thesis, we present two data hiding techniques in compression domain. The first method is called least-significant-bit substitution (LSBS) method. The basic idea of LSBS method is to analyze the significance of each bit of each coded frame, and then substitutes the LSBs with the data to be hidden. The second method is called dither-like data hiding (DDH) method, which utilizes the characteristics of subtractive dithering and the multistage vector quantization (MSVQ) in G.729 and MELP. The secret data is hidden in the index of the MSVQ. The data stream processed by either LSBS method or DDH method is compatible with MELP or G.729 speech coding standard. From the simulation results, both methods can deliver the secret message and the speech signal simultaneously, and the DDH method provides better quality than LSB Substitution Method at the same data embedding rate.
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Lo, Chun-Hung, and 駱俊宏. "Design and Implementation of an Efficient G.729 Voice Encoder." Thesis, 2010. http://ndltd.ncl.edu.tw/handle/e45x86.
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碩士<br>國立臺北科技大學<br>電腦與通訊研究所<br>98<br>As the Internet technology is progressing, the internet has been able to transmit voice, video and data in the existing network bandwidth. G.729 speech codec is set by the International Telecommunications Association (ITU-T). The compression standard is used on behalf of the Code Excited Linear Prediction (Conjugate-Structure Algebraic Code-Excited Linear-Prediction, CS-ACELP) coding algorithm. The G.729 speech coding is mainly used in wireless communication systems, digital satellite systems and digital private line. In this thesis, we use the reconfigurable architecture to implement the G.729 voice codec. We use the software oriented hardware-software co-design method to implement and evaluate the G.729 voice codec. The experimental results show that this work can reduce 36.26% system hardware resource and 44.52% system execution time.
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Wang, Jia-Yu, and 王嘉宇. "A Study on Fast Coding Algorithm for ITU-T G.723.1 and G.729 Speech Codecs." Thesis, 2010. http://ndltd.ncl.edu.tw/handle/72260792424405854470.
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碩士<br>南台科技大學<br>資訊工程系<br>98<br>Speech communication is the most common service in the Internet telecommunication and multimedia process. However, since speech signal should be continuously sent back, the voice in the service of the Internet should collect enough speech data, which can cause large speech delay and can degrade the speech quality in a limited network bandwidth. To achieve "continuity", speech codec with high compression rate has been used to generate a low-rate data stream, but that codec requires higher computational complexity. Thus, reducing the bit rate and improving speech quality of codec is the most significant. ITU-T offers the G.723.1and G.729 codecs that have used popularly in the Internet applications. These codecs offer high quality and low bit rate coding constitution.
This paper predict the search range of adaptive codebook-gain in the G.723.1 standard codec by minimizing the mean square error between the three-tap excitation signal with its residual signal and one-tap pitch predictor. For the G.723.1 MP-MLQ, we propose a fast search algorithm by using a designed energy function and the multi-track positions structure of the stochastic excitation signals to predict the candidate pulses for each subframe. As for both of the G.723.1 and the G.729 ACELP codebook, we base on depth-first tree search (DFS) and pulse-position likelihood-estimate to propose a fast search algorithm. As the two encoders belong to CELP coding structure, transcoding procedures are completed through two processes: line spectral pair and pitch conversions. They are all used to linear interpolation processing. For further computational complexity reduction, we use two fast search algorithms. First, we employ residual signals to predict candidate gain-vectors of adaptive-codebook in the G.723.1. Next, we adopt fast stochastic excitation pulses search method. Simulation results show that the proposed methods reduce a large amount of computation. Also, reconstructed speech signal still maintain a certain level of speech quality with perceptually negligible degradation.
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Chang, Sheng-Chih, and 張勝致. "The research of computation improvement on G.729’s fixed codebook." Thesis, 2007. http://ndltd.ncl.edu.tw/handle/99q95n.
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碩士<br>國立臺北科技大學<br>電機工程系研究所<br>95<br>The computation improvement on G.729’s fixed codebook is researched and implemented. Moreover then 20% computation of G.729 is paid on the fixed codebook search. In this thesis, an preprocessing algorithm is employed to limit the search domain of fixed codebook’s space. In the experimental results, more than 98.54% of computation is reduced. The speech quality is still good in this approach. The MOS of original G.729 is 4.5 and the new approach is about 3.8. It is reasonable good and the performance of this approach is confirmed.
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Tsai, Yu Ruei, and 蔡毓叡. "Complexity Reduction for Stochastic-Codebook Coding of G.729 Speech Coders." Thesis, 2008. http://ndltd.ncl.edu.tw/handle/01419377049601453671.
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碩士<br>南台科技大學<br>資訊工程系<br>96<br>The algebraic code excited linear prediction (ACELP) algorithm due to low complexity and high quality in its analysis-by-synthesis optimization has been adopted by many speech coding standards. This thesis studied the stochastic-codebook search of ACELP speech coder, since it is an important component for the quality of the coded speech. In order to reduce the computational complexity of ACELP coders, the focused search, the depth-first tree search and the global pulse-replacement search are suggested in the ITU-T G.729 G.729A and G.729.1, respectively. Based on the global pulse-replacement (GPR) search, this thesis proposed several derivative GPR approaches to reduce the computational complexity of stochastic-codebook search procedure. In addition, the switching search schemes are proposed to reduce the computation required for the ACELP optimization of ITU-T G.729D speech coder. By using two search rounds and limiting the search range, the computational complexity of the proposed approach is only 6.25% of the full search method recommended by G.729D. Simulation results show that the coded speech quality evaluated by using the standard subjective and objective quality measurements is with perceptually negligible degradation.
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Chen, Ching-Chang, and 陳慶彰. "A G.729 and G.723.1 Based Multi-Channel Speech Mixing Method for Multi-Point Conferencing System." Thesis, 2001. http://ndltd.ncl.edu.tw/handle/84681624107890309336.
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碩士<br>國立中央大學<br>電機工程研究所<br>89<br>Abstract
In a multi-point conference, users are offered a substitute for a face-to-face meeting within the economic constraints of the technology available. In this situation, an audio mixing scheme is needed to make the meeting successful. Audio mixing can create a full-duplex conversation environment that users can speak at any moment. Furthermore, it can be used in entertainment applications, such as audio chat rooms and online games.
Full decoding method is an intuitive and traditional audio mixing method, but it requires high computational complexity and long processing time. In this work, we propose a partial decoding method based on CELP coding architecture. This method selects a target frame as the mixed output from all incoming frames. There is no need for any encoding and decoding processes. Partial decoding method can be directly applied to CELP based speech coding, such as G.729 and G.723.1 speech standards. It achieves excellent voice quality as the full decoding method does while it only requires 5% to 8% computation loading.
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Hsun, Chen Shih, and 陳世勳. "The Implementation of G.729 Speech Coding on a 16-bit DSP Chip." Thesis, 2000. http://ndltd.ncl.edu.tw/handle/14468261541636374465.
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碩士<br>國立海洋大學<br>電機工程學系<br>88<br>The International Telecommunications Union－Telecommunications Standardization Sector (ITU-T) has already standardized conjugate structure－algebraic code excited linear prediction (CS-ACELP) that has toll quality at 8 kb/s as an advanced speech coding technology in November 1995. The ITU recommendation is known as G.729. The 8 kb/s CS-ACELP meets the requirement for low bit rate, short-delay time and high quality of synthesized speech signal.
In this thesis, the standards of G.729 speech coder is studied and the real time decoder is recoded partially by the assembly language of ADSP-2181 16 bits fixed-point DSP (Digital Signal Processor) chip to speed up its running time by a factor of 70. We also verified the complexity of subroutine of G.729 speech coder and illustrated the architecture and characteristic of DSP chip.
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Deshpande, Murali Mohan. "Speech Enhancement By Bandwidth Extension - A Codebook Based Approach In G.729 Compressed Domain." Thesis, 2004. http://etd.iisc.ernet.in/handle/2005/1165.
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Lin, Zhi-Zhong, and 林志忠. "An optimal Study on G.729-A Speech Coding Algorithm Using Fixed Point DSP TMS320C62x Processor." Thesis, 2002. http://ndltd.ncl.edu.tw/handle/22901574692115533427.
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碩士<br>國立高雄第一科技大學<br>電腦與通訊工程所<br>90<br>Low bit-rate and acceptable speech quality are very important factors for measuring the performance of a speech coder. The speech coder of Conjugate Structure Algebraic Code Excited Linear Prediction(CS-ACELP) is a speech compression scheme satisfied the above two conditions. This kind of the speech coder can be applied in International Mobile Telecommunication (IMT)-2000, Personal Communication System (PCS), and low Carrier-to-Noise(C/N) digital satellite system. This thesis presents the implementation of the CS-ACELP which has been chosen as G.729-A on the fixed-point 32 bits TMS320C6201 DSP processor. In the future, The IMT-2000 not only need to process speech but also multimedia . Therefore, we select the characteristics of more clocks and multi-operation units on the fixed-point DSP C6201 as a target to implement the G.729-A vocoder.
In the first stage, the Code Composer Studio(CCS) simulation environment is used to develop the standard CS-ACELP algorithm. Figuring out possible methods and techniques which can reduce the complexity of computation under the requirements of acceptable subject listening quality and 8kbps bit-rate. We apply these methods and techniques we obtained to the C6201 fixed point DSP Processor EVM .
We mainly use C language to develop our algorithm. And we concentrate on how to implement every module efficiently by utilizing the C6201 pipeline architecture and the building-in assembly language . In addition, the problems of overflow, accumulative error, and Q format modification are also the key issue in our research.
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Lin, Chiun-Chau, and 林群超. "A New VoIP Technique Combining Speech Data Encryption/G.729 Error Recovery and Its Integration with a Networked-Based Video/Speech Surveillance System." Thesis, 2003. http://ndltd.ncl.edu.tw/handle/00609153343994637179.
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