• The Journal of Korean Institute of Communications and Information Sciences
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• v.11 no.4
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• pp.287-295
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• 1986

In this paper, a SBC(sub-band-coding) is proposed to code a speech signal for a digital mobile radio and a robustness of speech quality of the SBC over the Rayleigh fading channel is investigated via a computer simulation. First the Rayleigh fading channel and 16-ary DPSK receiver models are presentes and verified its validitties by comparing with theoretical values. Three different measures: SNR, LPC distance measure and subjective listening test, were used to evaluate the effects due to the Rayleigh fading channel errors. From the results of computer simulation at BER=$10_{-3}$, $10_{-2}$, 5$ imes$$10_{-2}$, it was found that the speech remained quite intelligible at BER=$10_{-2}$and the link is still usuable even at BER=5$ imes$$10_{-2}$ Thus it was concluded that the SBC can be applicable to the digital mobile radio on the Rayleigh fading channel error in the range of $10_{-4}$~$10_{-2}$ without emplowing any error correction codes.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.3
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• pp.119-124
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• 2009

In this paper, AMR-WB+ audio coder is implemented, in real-time, using Intel 400MHz Xscale PXA250 with 32bit RISC processor ARM9E-J(R)core. The assembly code for ARM9E-J(R)core is developed through the serial process of C code optimization, cross compile, assembly code manual optimization and adjusting the optimized code to Embedded Visual C++ platform. C code is trimmed on Visual C++ platform. Cross compile and assembly code manual optimization are performed on CodeWarrior with ARM compiler. Through these stages the code for both ARM EVM board and PDA is implemented. The average complexities of the code are 160.75MHz on encoder and 33.05MHz on decoder. In case of static link library(SLL), the required memories are 65.21Kbyte, 32.01Kbyte and 279.81Kbyte on encoder, decoder and common sources, respectively. The implemented coder is evaluated using 16 test vectors given by 3GPP to verify the bit-exactness of the coder.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.7A
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• pp.1046-1051
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• 1999

It is well known that Parallel Concatenated Convolutional Codes(turbo codes) has a good performance for long block sizes. This thesis has analyzed the performance of turbo code which is based on voice or control frames with short frame sizes in the future mobile communication system. Also, at the similar decoding complexity, the performance of turbo code and convolutional codes in the speech/control frames, and the applicability of this system are considered. As a result, turbo code in short frame sizes present the performance of a BER of $10^{-3}$ or more over 3 iterations in the future mobile communication system. However, at a BER of $10^{-3}$ , if the same complexity is considered, the performance of rate 1/2 turbo code with K = 5 is better than that of convolutional code with K = 9 at low $E_b/N_0$, and the performance of turbo code with K = 3 is superior to that of convolutional code with K = 7. Rate 1/3 turbo code with K = 3 and 5 have similar to performance of rate 1/2 turbo code.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.38 no.3
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• pp.304-315
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• 2001

In this paper, we peformed the real-time implementation of AMR(Adaptive Multi-Rate) speech coding algorithm which is adopted for IMT-2000 service using TMS320C6201, i.e., a Texas Instrument´s fixed-point DSP. With the ANSI C source code released from ETSI, optimization is performed to make it run in real-time with memory as small as possible using the C compiler and assembly language. Implemented AMR speech codec has the size of 32.06 kWords program memory, 9.75 kWords data RAM memory, and 19.89 kWords data ROM memory. And, The time required for processing one frame of 20 ms length speech data is about 4.38 ms, and it is short enough for real-time operation. It is verified that the decoded result of the implemented speech codec on the DSP is identical with the PC simulation result using ANSI C code for test sequences. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.2 s.302
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• pp.131-142
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• 2005

The low rate speech coders under 4 kbit/s are based on sinusoidal transform coding (STC) or multiband excitation (MBE). Since the harmonic coders are not efficient to reconstruct the transient segments of speech signals such as onsets, offsets, non-periodic signals, etc, the coders do not provide a natural speech quality. This paper proposes method of a efficient transient model :d a multi-mode low rate coder at 2.4 kbit/s that uses harmonic model for the voiced speech, stochastic model for the unvoiced speech and a model using aperiodic pulse location tracking (APPT) for the transient segments, respectively. The APPT utilizes the harmonic model. The proposed method uses different models depending on the characteristics of LPC residual signals. In addition, it can combine synthesized excitation in CELP coding at time domain with that in harmonic coding at frequency domain efficiently. The proposed coder shows a better speech quality than 2.4 kbit/s version of the mixed excitation linear prediction (MELP) coder that is a U.S. Federal Standard for speech coder.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.5
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• pp.74-80
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• 2010

Speech and audio codecs have been developed based on different type of coding technologies since they have different characteristics of signal and applications. In harmony with a convergence between broadcasting and telecommunication system, international organizations for standardization such as 3GPP and ISO/IEC MPEG have tried to compress and transmit multimedia signals using unified codecs. MPEG recently initiated an activity to standardize the USAC (Unified speech and audio coding). However, USAC RM (Reference model) software has been problematic since it has a complex hierarchy, many useless source codes and poor quality of the encoder. To solve these problems, this paper introduces a new RM software designed with an open source paradigm. It was presented at the MPEG meeting in April, 2010 and the source code was released in June.

• Journal of the Korean Institute of Educational Facilities
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• v.11 no.1
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• pp.5-14
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• 2004

Since 1990s in Korea, elementary school classrooms have been designed toward open education system in pursuit of variety of educational purpose. Also, the architectural designs of schools have been acomplished for individual school not based on the standard design code. The present paper aims to investigate the acoustic environment of existing classrooms and to compare the sound insulation capacity between the ordinary classrooms and the newly built classrooms for open education. The current acoustical situation of elementary classrooms was analyzed using field measurements and questionnaire survey. In order to this, Three elementary schools were selected which were built in 1978, 1996 and 2000 respectively. Room acoustical parameters including Reverberation time(RT), Definition(D50), Speech Intelligibility(RASTI), Transmission loss(TL) and STC were measured in a classroom in each elementary school classroom. Each measurement was undertaken with the windows and doors being open or closed. As the result, it was found that the transmission loss between rooms in open classrooms is, $5{\sim}6dB$ in average, inferior than the ordinary classrooms. The RASTI of 0.70 was measured in newly built classrooms which is better than old classrooms(0.70) and open classrooms(0.73). This was shown as same in the speech definition measurements. This results from the condition of sealing and airtightness of classrooms and floor materials. The results denote that open classrooms have poor acoustic condition in sound insulation and speech intelligibility.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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• pp.899-903
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• 1994

This paper describes two methods to assess the output speech, quality of vocoders for telephone conferencing in digital mobile communication networks. The proposed methods are the sentence discrimiantion method and the modified degraded mean opinion score (MDMOS) test. We apply these two methods to Qualcomm code excited linear prediction (QCELP), vector sum excited linear prediction (VSELP) and regular pulse excited-long term predictin (RPE-LTD) voceders to evaluate which vocoding algorithm can process mixed voice signal from two speakers better for telephone conferencing. From the experiments we obtain that the VSELP vocoding algorithm reveals superior output speech quality to the other two.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06c
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• pp.214-217
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• 1994

The major drawback in the Code Excited Linear Prediction type vocoders is their large computational requirements. In this paper, a simple method is proposed to reduce the pitch searching time in the pitch filter almost without degradation of quality. On the basis of the observational regularity of the correlation function of speech, only the limited numbers of pitch lags are considered to be an optimum pitch. This is done by skipping the negative envelope side of the correlation function and limiting the maximum number of lags to be considered preliminarily. By doing so, we can reduce the computational time of pitch searching more than 51% with negligible quality degradation. In addition to that, by combining that method with the conventional delta search technique, we can reduce the computational time requirements more than 60% without serious lowering the speech quality in segmental SNR measure compared to the conventional full search method.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.51-54
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• 2000

This paper describes the fixed-point DSP implementation of a CELP(code-excited linear prediction)-based speech coder. The effective realization methodologies to maximize the utilization of the DSP's architectural features, specifically Parallel movement and pipelining are also presented together with the implementation results targeted for the ITU-T standard G.723.1 using Motorola DSP56309. The operation of the implemented speech coder is verified using the test vectors offered by the standard as well as using the peripheral interface circuits designed for the coder's real-time operation.