• The Transactions of The Korean Institute of Electrical Engineers
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• v.62 no.12
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• pp.1750-1754
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• 2013

A variable bit rate speech coder is presented. The coder is based on the observation that a speech signal can be viewed as a combination of piecewise linear signals in a short time period. The encoder detects the sample points where the slope of the signal changes, which are called the inflection points in this paper. The coder transmits the location and value for the detected inflection sample, but only the location information for the noninflection samples. In the decoder, the noninflection samples are estimated with interpolation of the received information. Several factors affecting the performance of the coder have been tested through simulation. Simulation results show that the linear interpolation produces 1 ~ 5 dB improvement over the cubic spline interpolation. And the -law companding does not provide any benefit when it is applied before the inflection detection. With low threshold values in the inflection point detection, the coder shows better MOS and more than 16 dB improvement in SNR compared to the continuously variable slope delta modulation (CVSDM).

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.10
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• pp.72-79
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• 1996

We developed a speech coder that can transfer data as well as speech for CDMA digital cellular system. We describe the design method of the speech coder that uses QCELP algorithm for speech coding. The speech coder is implemented on a single fixed-point DSP chip (TMS320C50). the coder has the complexity such as 4K words in RAM, 10K words in ROM, and 33 MIPS in execution time. The developed speech coder is fully tested and successfully working on the CDMA base station system.

• The Journal of the Acoustical Society of Korea
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• v.19 no.7
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• pp.42-47
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• 2000

This paper describes a real-time implementation of a GSM-EFR (Global System for Mobil communications Enhanced Full Rate) speech coder using OakDSP core; a 16bit fixed-point Digital Signal Processor (DSP) by DSP Group, Inc. The real-time implemented speech coder required about 24MIPS for computation and 7.06K words and 12.19K words for code and data memory, respectively. The implemented GSM-EFR speech coder passes all of test vectors provided by ETSI (European Telecommunication Standard Institute), and perceptual speech quality measurement using MNB algorithm shows that the quality of the GSM-EFR speech coder is similar to the one of 32kbps ADPCM. The real-time implemented GSM-EFR speech coder which is the highest bit-rate mode of the GSM-AMR speech coder will be used as the basic structure of the GSM-AMR speech coder which is embedded in MODEM ASIC of IMT2000 asynchronous mode mobile station.

• MALSORI
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• no.54
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• pp.27-43
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• 2005

Existing standard speech coders can provide speech communication of high quality while they degrade the performance of speech recognition systems that use the reconstructed speech by the coders. The main cause of the degradation is that the spectral envelope parameters in speech coding are optimized to speech quality rather than to the performance of speech recognition. For example, mel-frequency cepstral coefficient (MFCC) is generally known to provide better speech recognition performance than linear prediction coefficient (LPC) that is a typical parameter set in speech coding. In this paper, we propose a speech coder using MFCC instead of LPC to improve the performance of a server-based speech recognition system in network environments. However, the main drawback of using MFCC is to develop the efficient MFCC quantization with a low-bit rate. First, we explore the interframe correlation of MFCCs, which results in the predictive quantization of MFCC. Second, a safety-net scheme is proposed to make the MFCC-based speech coder robust to channel error. As a result, we propose a 8.7 kbps MFCC-based CELP coder. It is shown from a PESQ test that the proposed speech coder has a comparable speech quality to 8 kbps G.729 while it is shown that the performance of speech recognition using the proposed speech coder is better than that using G.729.

• Proceedings of the IEEK Conference
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• 2000.11d
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• pp.171-174
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• 2000

In this paper, We try to design combined source-channel coder that is compatible with video coding standards. This MAP decoder is proposed by adding semantic structure and semantic constraint of video coding standards to the method using redundnacy of the MAP decoders proposed previously. Then, We get the better performance than usual channel coder＇s.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.2
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• pp.88-94
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• 1996

In this paper, we present a half rate speech coder using trellis excitation. The coder combines code-excited linear prediction (CELP) system and trellis quantization method using the codebook expansion, and it produces higher speech quality than the typical CELP coder for the same transmission rate. A subjective comparison with 3~8 bit .$\mu$-law PCM indicates that the half rate coder provides speech quality between 5-bit and 6-bit $\mu$-law PCM .

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.510-521
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• 2002

MBE (multi-band excitation) coder can achieve high qualify synthetic speech below 4.0 kbps. There are, however, significant differences of the fine structure between the original spectrum and the synthetic spectrum. They are mainly due to the exclusive partition of voiced and unvoiced regions in frequency domain and the decision procedure based on the experimental threshold. This paper proposes MMBE (mixed multi-band excitation) speech model to overcome drawbacks of a MBE coder. In addition, two analysis methods, which do not need my decision procedure based on a threshold, are presented. Both voiced and unvoiced components can be mixed over all the frequency axis in the MMBE speech model. To illustrate the potential of the proposed speech model, we develop a 2.6 kbps MMBE coder and compare it with a 2.9 kbps MBE coder by both objective and subjective methods. The results have shown that the proposed coder has a better performance even at a lower bit-rate compared with the MBE coder.

• The Journal of the Acoustical Society of Korea
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• v.22 no.6
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• pp.443-451
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• 2003

Since a codebook-based CELP coder models its excitation signal according to one of several bit rates pre-assigned to codebooks and synthesizes speech signal using codebooks, it can not support encoding of speech signal at an arbitrary bit rate in one encoder. The proposed variable bit rate speech coder encodes the excitation signal based on the bit rate assigned to a present frame of speech using one-dimensional SPIHT and wavelet transform. Also it does't need to model excitation signal (or codebook) to some types as CELP coder, and can encode excitation signal at various bit rates without exact pitch information according to user requirement. As a result, since the coder doesn't have a codebook structure, it has relatively low coder complexity and provides equal or better speech quality compared to G.729 and G.723.1 coder.

• MALSORI
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• v.55
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• pp.143-158
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• 2005

The adequate compression algorithm is essential to achieve high quality embedded TTS system. in this paper, we Propose waveform interpolation coder for TTS corpus compression after many speech coder investigation. Unlike speech coders in communication system, compression rate and anality are more important factors in TTS DB compression than other performance criteria. Thus we select waveform interpolation algorithm because it provides good speech quality under high compression rate at the cost of complexity. The implemented coder has bit rate 6kbps with quality degradation 0.47. The performance indicates that the waveform interpolation is adequate for TTS DB compression with some further study.

• Proceedings of the IEEK Conference
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• 1998.06a
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• pp.19-22
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• 1998

In this study, the efficient channel coding method combined with CS-ACELP is proposed. The same convolutional coder and Viterbi decoder of COMA mobile communication system is used as channel coder. To make the best available use of limited channel coding redundancy, unequal error protection of punctured convolutional coder is used for variable reate allocation. But, the overall code rate is given by 2. The performance of proposed coder is analyzed and simulated in a Rayleigh fading channel. Experimental results show that the objective and subjective speech quality of variable rate channel coding methods are superior to those of non-variable channel coding method.