• 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 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.

• 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 the Acoustical Society of Korea
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• v.21 no.6
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• pp.531-542
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• 2002

Novel tandemless transcoding method for AMR and EVRC speech coders is proposed in this paper. In contrast to conventional tandem method, the parameters which is used commonly in speech coder where CELP algorithm is adapted are directly transcoded. The proposed algorithm is composed of LSP transcoding, pitch delay transcoding, gains transcoding and fixed codebook vector transcoding Evaluation results show that the novel algorithm achieves better speech quality than tandem method and reduce computational complexity and delay.

• The Journal of the Acoustical Society of Korea
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• v.18 no.6
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• pp.38-43
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• 1999

The EVRC vocoder is a toll quality coder, but it shows significant degradation or the quality in weak RF environment. In this paper, the speech quality degradation phenomenon of the EVRC is analyzed, and two methods are proposed as the solution - the pitch prediction and the gradual increase. The preference tests for various Rf environment are performed for speech quality assessments and both the methods show better performance.

• 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.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.169-172
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• 1999

본 논문에서는 EVRC 음성 부호화 시스템을 $OakDSPCore^{\circledR}$를 기반으로 설계된 C&S Technology사의 CSD17C00 칩을 이용하여 전 과정을 어셈블리어로 실시간 구현하였다. 구현된 EVRC 음성 부호화기는 최대의 계산량을 요구하는 8kbps일때 잡음제거 알고리즘을 제외한 인코더부분이 평균 22.5MIPS 이며, 디코더부분은 약 3.35MIPS의 복잡도를 나타낸다. 사용된 메모리양은 프로그램 ROM 10.8K words 데이터 ROM(table) 6.72K words 및 RAM 2.94K words이다. 구현된 EVRC 음성 부호화기는 북미 표준화 기구인 TIA(Telecommunications Industry Association)에서 제공하는 19 개의 test 백터들을 모두 통과하였다.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.751-754
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• 2000

본 논문에서는 EVRC 음성 부호화 시스템을 OakDSPCore(R) 를 기반으로 설계된 C&S Technology사의 CSD17C00칩을 이용하여 전 과정을 어셈블리어로 실시간 구현하였다. 구현된 EVRC음성 부호화기는 최대의 계산량이 요구되는8kbps 전송 모드일 때, 인코더부분이 최대24.45MIPS 이며 디코더부분은 3.35MIPS의 복잡도를 나타낸다. 사용된 메모리양은 프로그램 ROM 12.2Kworsd, 데이터 ROM(table) 6.72Kwords 및 RAM2.94Kwords 이다. 구현된 EVRC음성 부호화기는 북미 표준화 기구인Telecommunications Industry Association(TIA)에서 제공하는 19개의 test 벡터들을 모두 통과하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.129-132
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• 2001

본 논문에서는 EVRC 음성 부호화 시스템을 7MS320C54x를 이용하여 전 과정을 어셈블리어로 실시간 구현하였다. 구현된 EVRC 음성 부호화기는 최대의 계산량을 요구하는 8kbps 일때 잡음제거 알고리즘을 제치한 인코더부분이 평균 26.3MIPS 이며 디코더부분은 3.6MIPS의 복잡도를 나타낸다. 사용된 메모리양은 프로그램 ROM 16.9K words, 데이터 ROM(table) 7.9K words 및 RAM 5.8K words 이다. 구현된 EVRC 음성 부호화기는 북미 표준화 기구인 TIA(Telecommunications Industry Association)에서 제공하는 19 개의 테스트 백터들을 모두 통과하였다.

• The Journal of the Acoustical Society of Korea
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• v.26 no.6
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• pp.301-305
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• 2007

In this paper, we propose a simple algorithm for detecting underwater transient signals on the fact that the frequency range of underwater transient signals is similar to audio frequency. For this, we use a preprocessing module of EVRC speech codec that is the standard speech codec of the mobile communications. If a signal is entered into EVRC noise suppression module, we can get some parameters such as the update flag, the energy of each channel, the noise suppressed signal, the energy of input signal, the energy of background noise, and the energy of enhanced signal. Therefore the energy of the enhanced signal that is normalized with the energy of the background noise is compared with the pre-defined detection threshold, and then we can detect the transient signal. And the detection threshold is updated using the previous value in the noisy period. The experimental result shows that the proposed algorithm has $0{\sim}4% error rate in the AWGN or the colored noise environment.