• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.43-49
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• 2001

EVRC (Enhanced Variable Rate Codec) has been adopted as a standard coder for the CDMA digital cellular system in North America and Korea, and known to provide good call quality at 8kbps. In this paper, fast implementation algorithms for EVRC encoder are proposed. The proposed algorithms are based on both efficient pitch detection scheme and fast fixed codebook search algorithm. In the codebook search, computational complexity is reduced down to 70％ of the original EVRC by limiting the number of pulse position combination and by using a truncated impulse response. The proposed algorithms enable us to implement the EVRC with much smaller computational works. Also, informal subjective tests confirmed that the difference in the speech quality between the original EVRC and the proposed method was indistinguishable.

• The Journal of the Acoustical Society of Korea
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• v.21 no.2
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• pp.167-172
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• 2002

The EVRC (Enhanced Variable Rate Codec) is currently in service as a speech cosec in digital cellular systems in North America and Korea. In the EVRC, the LSP (Line Spectral Pairs) related to energy distribution of speech signals in the frequency domain are coded by weighted split vector quantization. Considering that the LSP codebooks might be trained with the language of the develop country of the codebooks or English, it is expected that codebooks trained with Korean provide the performance improvements in the communication in Korean. In this paper, the EVRC LSP codebooks are designed with korean adopting the LBG algorithm based vector quantization, and the performance improvement of the vector quantization and the accompanying speech quality improvement are demonstrated by spectral distortion, SNR and SegSNR measurements, respectively.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5C
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• pp.476-485
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• 2003

This paper proposers a preprocessing approach to improving the quality of the music produced by the EVRC(enhanced variable rate codec) which is one of the CDMA(Code Division Multiple Access) voice codecs. Since the EVRC is optimized only for speech signals, it can deteriorate the quality of the music passed through it. One of the problems with the EVRC-coded music is time-clipping, which usually occurs when subsequent frames are encoded at Rate l/8. Since the EVRC determines the bit rate for an input frame based on the long-term prediction gain, we increase the long-term prediction gain in order for the most of the frames to be encoded at Rate 1 or Rate 1/2. Experimental results show that the approach works well on music signals and the number of time-clipped frames is considerably reduced.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.8 no.3
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• pp.563-568
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• 2007

We propose the computation reduction method of real root method that is used in the EVRC(Enhanced Variable Rate Codec) system. The real root method is that if polynomial equations have the real roots, we are able to find those and transform them into LSP. However, this method takes much time to compute, because the root searching is processed sequentially in frequency region. But, the important characteristic of LSP is that most of coefficients are occurred in specific frequency region. So, to reduce the computation time of real root, we used the met scale that is linear below 1kHz and logarithmic above. In order to compare real root method with proposed method, we measured the following two. First, we compared the position of transformed LSP(Line Spectrum Pairs) parameters in the proposed method with these of real root method. Second, we measured how long computation time is reduced. The experimental result is that the searching time was reduced by about 48% in average without the change of LSP parameters.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.43 no.3 s.309
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• pp.54-60
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• 2006

This paper presents an effective algorithm for transcoding between the Enhanced Variable Rate Codec(EVRC) and G.729A. The simplest way to communicate between heterogeneous speech networks is the cascade connection of two different codecs, called tandem coding. However, tandem coding not only produces high computational loads, but also makes long delay, These problems can be solved by using the transcoding algorithm. The proposed algorithm consists of LSP (Line Spectral Pair) conversion, pitch delay conversion and algorithm for reduction of delay. Experimental results show the proposed algorithm produces lower computational complexity, shorter algorithm delay, and similar speech quality when compared with the tandem algorithm.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.12 no.12
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• pp.5843-5848
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• 2011

To improve the efficiency of the channel spectrum and to reduce the power consumption of the system in EVRC, the voice signal is compressed and transmitted only when the user speaks to. In addition to this, voice frames are divided into three rates 1, 1/2 and 1/8 and each frame is handled differently. For example, we assumed that the input is silence region if the 1/8 rate is used. In this paper, the sections are firstly separated into the voiced speech signal region, unvoiced speech signal region, and silence region by using distribution characteristics of LSP parameters. Then the paper suggested to encode 1 rate for the voiced speech signal, 1/2 rate for the unvoiced speech signal region, 1/8 rate for the silence region. In other words, traditional way of transmission is used when sending full rate in the EVRC. However, when sending half rate, the voice is firstly distinguished between voiced and unvoiced. If the voice is distinguished as voiced, voice is converted into full rate before the transmission. If it is distinguished as silence, EVRC's basic rate is applied. In the experimental results with SNR, ASDM, transmission bit rate measurement, we have demonstrated that voice quality was improved by using the proposed algorithm.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.696-708
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• 2002

In this paper, a novel transcoding algorithm for the Adaptive Multi Rate(AMR) and the Enhanced Variable Rate Codec(EVRC) vocoders via direct parameter transformation is proposed. In contrast to the conventional tandem transcoding algorithm, the proposed algorithm converts the parameters of one coder to the other without going through the decoding and encoding processes. The proposed algorithm consists of the parameter decoding, frame classification, mode decision, and transcoders for two frame types. The transcoders convert the parameters such as LSP, frame energy, pitch delay for the adaptive codebook, fixed codebook vector, and codebook gains. Evaluation results show that while exhibiting better computational and delay characteristics, the proposed algorithm produces equivalent speech quality to that produced by the tandem transcoding algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5C
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• pp.548-554
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• 2003

Interoperability is ole the most important factors for a successful integration of the speech network. To accomplish communication between endpoints employing different speech coders, decoder and encoder of each endpoint coder should be placed in tandem. However, tandem coder often produces problems such as poor speech quality, high computational load, and additional transmission delay. In this paper, we propose an efficient transcoding algorithm that can provide interoperability to the networks employing ITU-T G.723.1［1］and TIA IS-127 EVRC［2］speech coders. The proposed transcoding algorithm is composed of four parts: LSP conversion, open-loop pitch conversion, fast adaptive codebook search, and fast fixed codebook search. Subjective and objective quality evaluation confirmed that the speech quality produced by the proposed transcoding algorithm was equivalent to, or better than the tandem coding, while it had shorter processing delay and less computational complexity, which is certified implementing on TMS320C62x.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.13 no.2
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• pp.962-977
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• 2019

The human voice is a convenient method of information transfer between different objects such as between men, men and machine, between machines. The development of information and communication technology, the voice has been able to transfer farther than before. The way to communicate, it is to convert the voice to another form, transmit it, and then reconvert it back to sound. In such a communication process, a vocoder is a method of converting and re-converting a voice and sound. The CELP (Code-Excited Linear Prediction) type vocoder, one of the voice codecs, is adapted as a standard codec since it provides high quality sound even though its transmission speed is relatively low. The EVRC (Enhanced Variable Rate CODEC) and QCELP (Qualcomm Code-Excited Linear Prediction), variable bit rate vocoders, are used for mobile phones in 3G environment. For the real-time implementation of a vocoder, the reduction of sound quality is a typical problem. To improve the sound quality, that is important to know the size and shape of noise. In the existing sound quality improvement method, the voice activated is detected or used, or statistical methods are used by the large mount of data. However, there is a disadvantage in that no noise can be detected, when there is a continuous signal or when a change in noise is large.This paper focused on finding a better way to decrease the reduction of sound quality in lower bit transmission environments. Based on simulation results, this study proposed a preprocessor application that estimates the SNR (Signal to Noise Ratio) using the spectral SNR estimation method. The SNR estimation method adopted the IMBE (Improved Multi-Band Excitation) instead of using the SNR, which is a continuous speech signal. Finally, this application improves the quality of the vocoder by enhancing sound quality adaptively.