• 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.

• Journal of Broadcast Engineering
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• v.23 no.1
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• pp.86-92
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• 2018

In this paper, we propose a method for coding history detection of digital speech signal. In digital speech communication and storage, the signal is encoded to reduce the number of bits. Therefore, when a speech signal waveform is given, we need to detect its coding history so that we can determine whether the signal is an original or an coded one, and if coded, determine the number of times of coding. In this paper, we propose a coding history detection method for 12.2kbps AMR codec in terms of original, single coding, and double coding. The proposed method extracts a speech-specific feature vector from the given speech, and models the feature vector using a deep neural network. We confirm that the proposed feature vector provides better performance in coding history detection than the feature vector computed from the general spectrogram.

• Journal of the Korean Institute of Telematics and Electronics
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• v.27 no.6
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• pp.821-828
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• 1990

A forward error correction (FEC) technique is presented for speech data in 16 Kbps digital mobile communications. 14Kbps SBC-APCM(AQB) and QPSK are used as speech coding and modulation techniques, respectively. Because each bit in a speech data block had different importance, applying FEC to speech data bit-selectively in more effective than applying FEC to all speech data equally. To select bits in a speech data block to be protected by FEC the bit error sensitivity of each bit is computed. For a few BCH and Reed-Solomon codes used as bit-selective FEC the performance of the coding technique is computed.

• Journal of the Korean Institute of Telematics and Electronics
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• v.27 no.4
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• pp.491-501
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• 1990

Recently, four speech coding techniques, namely, SBC-APCM(sub-band coding adaptive PCM), RPE-LPC(regualr pulse excitation linear predictive codec), MPE-LTP(multi-pulse excited long-term prediction) and CELP (code-excited linear prediction) are proposed for digital mobile radio applications. However, a performance comparison of these coders in the Rayleigh fading environment has not been made yet. In this paper, the performances of the four spech coders in the random bit error and burst error environment are investigated. For the channel coding of SBC-APCM, RPE-LPC and MPE-LTP, the sensitivity of output bit stream is measured and a bit selective forward error correction is provided acording to the measured bit sensitivity. And for an attempt to improve the performance of CELP, an optimum quantizer is applied for transmitting scalar quantities in CELP. However, an improvement over the conventional approach is found to be negligible. For the channel coding of CELP, Reed-Solomon code, Golay code, convolutional code of rate 1/2 shows the best performance. Finally, from the simulation results, it is concluded that CELP is the best candidate for digital mobile radio and is followed by MPE-LTP, SBC-APCM and RPE-LPC.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.9
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• pp.1838-1847
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• 1994

Until recently, they decided the standard of the digital mobile communication speech coding method and competively developed the more detailed techniques in North America, Europe, Japan, etc. But, we have not yet determined. In this paper, we compared the RPE-LTP speech coding algorithm, standard in Europe, with the VSELP speech coding algorith, standard in North America, with respect to the soruce coding. We described the comprehensive verification and comparison with each speech coder, and discussed the improvement plan. Next, we also compared the number of computations which affects the real time processing seriously. Moreover, we performed the simulation with the Korean speech data, concreting the algorithm of each speech coder. Finally, we compared the performance of each speech coder with segmental SNR and 5-point MOS. The number of computations was calculated, and the result was that the number of multiplication computing times of VSELP speech encoder was the largest. With 26 speech data, the segmental SNR of VSELP was calculated larger than that of RPE-LTP. The 5-point MOS test was performed, and the result was that the basic speech quality of VSELP was equivalent or better than that of RPE-LTP.

• Journal of Digital Convergence
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• v.12 no.6
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• pp.271-276
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• 2014

In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech quality in case coexist with a voiced and unvoiced consonants in a frame. In this paper, I propose a method of 8kbps Multi-Pulse Speech Coding(FBD-MPC: Frequency Band Division MPC) by using TSIUVC(Transition Segment Including Unvoiced Consonant) searching, extraction and approximation-synthesis method in a frequency domain. I evaluate the 8kbps MPC and FBD-MPC. As a result, SNRseg of FBD-MPC was improved 0.5dB for female voice and 0.2dB for male voice respectively. Compared to the MPC, SNRseg of FBD-MPC has been improved that I was able to control the distortion of the speech waveform finally. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

• The Transactions of the Korea Information Processing Society
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• v.7 no.4
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• pp.1299-1304
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• 2000

In a speed coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech quality in case coexist with a voiced and an unvoiced consonants in a frame. So, I propose TSIUVC(Transition Segment Including UnVoiced Consonant) searching, extraction ad approximation-synthesis method in order to uncoexistent with a voiced and unvoiced consonants in a frame. This method based on a zerocrossing rate and pitch detector using FIR-STREAK Digital Filter. As a result, the extraction rates of TSIUVC are 84.8% (plosive), 94.9%(fricative), 92.3%(affricative) in female voice, and 88%(plosive), 94.9%(fricative), 92.3%(affricative) in male voice respectively, Also, I obain a high quality approximation-synthesis waveforms within TSIUVC by using frequency information of 0.547kHz below and 2.813kHz above. This method has the capability of being applied to speech coding of low bit rate, speech analysis and speech synthesis.

• The Journal of the Acoustical Society of Korea
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• v.18 no.6
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• pp.71-76
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• 1999

In digital cellular communication systems, tandem coding occurs in communications between mobile phones with different speech codecs, resulting in poor voice quality, high computational load, and long transmission delay. In this paper, voice packet conversion technique is proposed to solve the tandem coding problems, and packet conversion algorithm from 13kbps QCELP to 8kbps QCELP is developed. Simulations using various speech data show that the proposed packet conversion method produces voice quality which is equivalent to that by the conventional tandem coding method with shorter transmission delay using about 33% computational load.

• Journal of Digital Convergence
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• v.11 no.5
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• pp.285-290
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• 2013

In a MPC coding using excitation source of voiced and unvoiced, it would be a distortion of speech waveform. This is caused by normalization of synthesis speech waveform of voiced in the process of restoration the multi-pulses of representation section. To solve this problem, this paper present a method of position compensation(PC-MPC) in a multi-pulses each pitch interval in order to reduce distortion of speech waveform. I was confirmed that the method can be synthesized close to the original speech waveform. And I evaluate the MPC and PC-MPC using multi-pulses position compensation method. As a result, $SNR_{seg}$ of PC-MPC was improved 0.4dB for female voice and 0.5dB for male voice respectively. Compared to the MPC, $SNR_{seg}$ of PC-MPC has been improved that I was able to control the distortion of the speech waveform finally. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

• The Transactions of the Korea Information Processing Society
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• v.6 no.1
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• pp.247-252
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• 1999

In order to realize a speech coding at low bit rates, a pitch information is useful parameter. In case of extracting an average pitch information form continuous speech, the several pitch errors appear in a frame which consonant and vowel are coexistent; in the boundary between adjoining frames and beginning or ending of a sentence. In this paper, I propose an Individual Pitch (IP) extraction method using residual signals of the FIR-STREAK digital filter in order to restrict the pitch extraction errors. This method is based on not averaging pitch intervals in order to accomodate the changes in each pitch interval. As a result, in case of Ip extraction method suing FIR-STREAK digital filter, I can't find the pitch errors in a frame which consonant and vowel are consistent; in the boundary between adjoining frames and beginning or ending of a sentence. This method has the capability of being applied to many fields, such as speech coding, speech analysis, speech synthesis and speech recognition.