• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.394-400
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• 2002

In digital speech communications, the quality of service can be increased by speech coding scheme that is adaptive to the error rate of voice packet transmission. However, current communication protocol in cellular and internet communications does not provide the function that transmits the channel error information. To solute this problem, in this paper, new method for real-time transmission of channel error information is proposed, where channel error information is embedded in voice packet. The proposed method utilizes the pulse positions of codevector in ACELP speech codec, which results in little degradation in speech quality and low false alarm rate. The simulations with various speech data show that the proposed method meets the requirement in speech quality, detection rate, and false alarm rate.

• 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 Korean Society of Laryngology, Phoniatrics and Logopedics
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• v.19 no.2
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• pp.101-106
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• 2008

The speech signal has been put into a form suitable for storage and analysis by computer, several different operation can be performed. Filtering, sampling and quantization are the basic operation in digiting a speech signal. The waveform can be displayed, measured and even edited, and spectra can be computed using methods such as the Fast Fourier Transform (FFT), Linear predictive Coding (LPC), Cepstrum and filtering. The digitized signal also can be used to generate spectrograms. The spectrograph provide major advantages to the study of speech. So, author introduces the basic techniques for the acoustic recording, digital signal processing and the principles of spectrum and spectrogram.

• Journal of Broadcast Engineering
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• v.11 no.2 s.31
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• pp.200-206
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• 2006

This paper presents a preprocessing method to modify the input audio signals of a speech coder to obtain the finally enhanced signals at the decoder. For the purpose, we introduce the noise suppression (NS) scheme and the adaptive gain control (AGC) where an audio input and its coding error are considered as a noisy signal and a noise, respectively. The coding error is suppressed from the input and then the suppressed input is level aligned to the original input by the following AGC operation. Consequently, this preprocessing method makes the spectral energy of the music input redistributed all over the spectral domain so that the preprocessed music can be coded more effectively by the following coder. As an artifact, this procedure needs an additional encoding pass to calculate the coding error. However, it provides a generalized formulation applicable to a lot of existing speech coders. By preference listening tests, it was indicated that the proposed approach produces significant enhancements in the perceived music qualities.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.16 no.1
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• pp.85-100
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• 1991

In the discussion of digital mobile communication systems the speech coder based on the high quality low bit rate is an essential part of topics to overcome the limited availability of radio spectrum, which will enhance the communication services. In this paper we present the implementation and performance evaluation of 13kbps RPE LTP speech coder. An implementation of a real time full duplex coder with 75% of DSP loading rate using a single DSP chip has been shown, and also the fixed point simulations for H/W implementation has been performed. Finally, analysis result for relative bit importance of each transmitting parameter has been shown for channel coding.

• Journal of the Korean Institute of Telematics and Electronics
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• v.23 no.4
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• pp.429-438
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• 1986

In this paper we have designed and implemented a real-time, full-duplex SBC (sub-band coding) codec at 16kbps using a high speed digital signal processor, NEC 7720. The SBC codec employs a QMF(quadrature mirror filter) filter bank based on the tree structures of two-band analysis-synthesis pairs to partition speech signal into 4 octabe bands. Computer simulation has been done to investigate the effect of fixed-point computation of the NEC 7720. Three different performance measures, the conventional signal-to-noise ratio, the informal listening test, and an LPC(linear predictive coding)distance measure, have been used in this simulation. The necessary parameters have been optimized through the simulation. The developed hardware and software have been tested in real-time operation using a hardware emulator.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.679-682
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• 1999

This paper proposes the LSF or LSP that is the method of using to transfer the speech parameters after processed the speech to LPC, which is digital coding transferring efficiently, for the best quality and the lowest bit rate of parameters. The new revised transform algorithm between LSF and LPC coefficients is proposed. The proposed algorithm eliminates all multiplications, computes fewer operations, and reduces memory buffer sizes.

• The Journal of the Acoustical Society of Korea
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• v.14 no.1E
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• pp.30-38
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• 1995

In this paper, we evaluated the performance of vocoders for three-way conferencing using signal summation technique in digital mobile communication network. The signal summation technique yields natural mode of three-way conferencing, in shich the mixed voice signal from two speakers are transmitted to a third person, though there has been no useful speech coding technique for the mixed voice signal yet. We established Qualcomm code term prediction (RPE-LTP) vocoders to provide three-way conferencing using signal summation techinique. In addition, as the conventional speech quality measures are not applicable to the vocoders for mixed voice signals, we proposed two kinds of subjective quality measures. These are the sentence discrimination (SD) test and the modified degraded mean opinion score (MDMOS) test. The experimental results show that the output speech quality of the VSELP vocoder is superior to other two.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.8 no.3
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• pp.113-119
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• 1983

In this paper, a method of the step size adaption is studied on the delta modulation coding of speech signals. Exponential adaption processes are reserched by a new circuit model. It is presented a shorten error recovery in decoder step size. Practical considerations favor one algorithm, and its digital implementation has been adapted for the illustration of above method, using the rate multipliers and the validity is verified by laboratory experiment.