• The Journal of the Acoustical Society of Korea
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• v.28 no.2
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• pp.148-154
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• 2009

In audio coding, SBR technology synthesizes the high-bands using patched time-frequency information from low-bands and the correction parameters, Since SBR transmits only correction parameters for high-bands, it provides a low-rate coding of high-bands, and is used as a core module of MPEG-4 HE-AAC, SBR was originally designed for audio signal and its performance for speech signal tends to decrease, and the major reason is an excessive noise floor in high-bands which is caused by incorrect tonality computation, In this paper, a new method to determine noise floor level in an adaptive fashion according to the speech characteristics is proposed in order to solve the problem of SBR for speech signal, The proposed method maintains the compatibility with the standard SBR, and the subjective performance evaluation shows that the proposed method improves the SBR performance especially for male speech signal compared with the standard SBR.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.38 no.5
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• pp.576-582
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• 2001

The Multi-Band Excitation(MBE) speech coder uses a different approach for the representation of the excitation signal. It replaces the frame-based single voiced/unvoiced classification of a classical speech coder with a set of such decision over harmonic intervals in the frequency domain. This enables each speech segment to be a mixture of voiced and unvoiced, and improves the synthetic speech quality by reducing decision errors that might occur on the frame-based single voiced and unvoiced decision process when input speech is degraded with noise. The IMBE-LP, improved version of MBE with linear prediction, represents the spectral information of MBE model with linear prediction coefficients to obtain low bit rate of 2.4 kbps. In this Paper, we proposed a variable rate IMBE-LP vocoder that has lower bit rate than IMBE-LP without degrading the synthetic speech quality. To determine the LP order, it uses the spectral band information of the MBE model that has something to do with he input speech's characteristics. Experimental results are riven with our findings and discussions.

• Journal of Korea Multimedia Society
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• v.7 no.7
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• pp.896-904
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• 2004

In this paper, We present an efficient variable rate speech coder for spectral efficiency and packet processing technique for packet loss compensation of a voice codec with 10msec frame in VoIP service. Through disconnecting the users from the spectral resource during silence interval of about 60% period, a variable rate voice coder based on a voice activity detection(VAD) can increase spectral gain by two times. The performance of the method was analyzed by variation of detected voice activity factor and degraded speech frame ratio under various background noise level, and compared those of G.729B of ITU-T 8kbps standard speech codec. A method to compensate lost packets utilized addition of recovery data to a main stream and error concealment scheme for speech quality enhancement, the performance is verified by reconstructed speech quality. The proposed scheme can achieve spectral gain by two times or enhance speech quality by 3dB through reserved bandwidth of VAD. Therefore, the proposed method can enhance a spectral efficiency or speech quality of VoIP.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2001.11b
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• pp.109-113
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• 2001

최근 들어 디지털 이동통신 시스템은 음성 통신 뿐만 아니라 멀티미디어 통신에도 그 적용범위를 확장시키고 있다. 그러나 이러한 시스템에서 한정된 채널 대역폭의 사용은 멀티미디어 정보의 대용량성을 고려할 때, 그 적용범위를 심각하게 축소시키는 가장 큰 제한요소이다. 일반적인 통신 시스템의 소스 부호화기는 채널 잡음을 고려하지 않고 설계되며, 채널 부호화기는 소스 신호의 특성과 무관하게 채널 환경의 극복에만 중점을 두고 설계된다. 그러나 대역폭 제한적인 통신 환경에서 보다 효율적인 대역폭 사용을 위해서는 채널 환경에 따라 소스 부호율과 채널 부호율을 가변적으로 운용하여야 한다. 본 논문에서는 영상을 부대역 부호화하여 각 부대역 영상이 원래의 영상 재구성에 기여하는 중요도에 따라 한정된 채널 자원을 최적으로 할당하는 공동 소스/채널 부호화(Joint Source-channel Coding)에 관하여 연구하였다. 부대역 영상의 소스 부호화로는 DPCM과 PCM을 사용하였고, 채널 부호화는 TCM 부호화기를 사용하였으며 상이오류보호(Unequal Error Protection)를 위해 3가지 채널 부호율에 따라 각각 TCQPSK, TC8PSK, TC16PSK 변조방식을 사용하였다. 모의실험에 사용된 채널 환경은 랜덤잡음 환경과 이동수신의 경우에 발생하는 Rayleigh 페이딩 환경을 고려하였으며, 각 환경에서의 채널 SNR에 따라 동일오류보호(Equal Error Protection) 시스템과 상이오류보호(Unequal Error Protection) 시스템의 성능을 비교하였다.

• The Journal of the Acoustical Society of Korea
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• v.20 no.5
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• pp.35-47
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• 2001

This paper describes an efficient harmonic-CELP speech coder by taking advantages of harmonic and CELP coders into account. According to frame voicing decision, the proposed harmonic-CELP coder adopts the RP-VSELP coder as a fast CELP in case of an unvoiced frame, or an improved harmonic coder in case of a voiced frame. The proposed coder has main features as follows: simple pitch detection, fast harmonic estimation, variable dimension harmonic vector quantization, perceptual weighting reflecting frequency resolution, fast harmonic synthesis, naturalness control using band voicing, and multi-mode. These features make the proposed coder require very low complexity, compared with HVXC coder To demonstrate the performance of the proposed coder, a 2.4 kbps coder has been implemented and compared with reference coders. From results of informal listening tests, the proposed coder showed good quality while requiring low delay and complexity.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.105-108
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• 2004

음성 부호화 기술은 전송률과 복잡도를 줄이고 음질을 향상시키는 방향으로 진행되고 있다. 현재 상용화되고 있는 CELP형 보코더는 낮은 전송률에 비해 우수한 음질을 제공한다. 본 논문에서는 기존의 방식과 다르게 보코더 단에 입력 음성이 들어가기 앞서 전처리 기법을 수행하는 전처리단을 부가하여 전송률을 낮추는 방법을 소개하고, 소개된 방법들을 각기 비교하고 분석하고자 한다. 전처리기법들을 음성 인식이나 합성에서 사용되는 파라미터들을 적용시켰으며, 처리시간이나 계산시간에 있어 기존의 방식에서 많은 영향을 미치지 않은 간단한 알고리즘으로 구현하였다. 소개하는 전처리단에서는 기존의 코딩방식에서 사용하지 않은 파라미터들, 발성율, 지속시간, PSOLA 방식들을 이용하였다.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.18 no.1
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• pp.129-136
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• 2004

In this paper, we proposed quad variable rates ADPCM coding method for efficient speech transmission and real time porcessing is implemented on TMS320C6711-DSP. The modified ADPCM with four variable coding rates, 16[kbps], 24[kbps], 32[kbps] and 40[kbps] are used for speech window samples for good quality speech transmission at a small data bits and real time encoding and decoding is implemented using DSP. ZCR is used to identify the influence of the noise on the speech signal and to decide the rate change threshold. For noise superior signals, low coding rates are applied to minimize data bit and for noise inferior signals, high coding rates are applied to enhance the speech quality. In most speech telecommunications, silent period takes more than half of the signals, speech quality close to 40[kbps] can be obtained at comparabley low data bits and this is shown by simulation and experiments. TMS320C6711-DSK board has 128K flash memory and performance of 1333MIPS and has meets the requirements for real time implementation of proposed coding algorithm.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.62 no.4
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• pp.562-565
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• 2013

A new variable rate speech coding technique is proposed. The method is based on the observation that the speech signal approximately looks linear for a very short period of time. The information transmitted is the location and data value of inflection points. If the distance between the inflection points is large, the mid point location and its data value are also delivered. Thus, the encoder transmits both the location and the data value for the inflection samples, but the location only for the non-inflection points. The location information is expressed using one bit for each sample, 0 for non-inflection and 1 for inflection point. At the receiver, using the interpolation, the decoder estimates the untransmitted sample values for non-inflection locations from the received sample values for the inflection samples. With 50 % of computational cost of the existing CVSD delta modulation, the proposed method is expected to achieve the data rate of 36 to 38 kbps and the SNR of 10 to 13 dB.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.6
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• pp.84-89
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• 2003

In this paper, we real-time implemented to the TMS320C5416 the vocoder of variable bit rate applied the SOLA-B algorithm by Henja to the ITU-T G.729A vocoder of 8kbps transmission rate. This proposed method using the SOLA-B algorithm is that it is reduced the duration of the speech in encoding and is played at the speed of normal by extending the duration of the speech in decoding. At this time, we bandied that the interval of cross correlation function if skipped every 3 sample for decreasing the computational amount of SOLA-B algorithm. The real-time implemented vocoder of C.729A and SOLA-B algorithm is represented the complexity of maximum that is 10.2MIPS in encoder and 2.8MIPS in decoder of 8kbps transmission rate. Also, it is represented the complexity of maximum that is 18.5MIPS in encoder and 13.1MIPS in decoder of 6kbps, it is 18.5MIPS in encoder and 13.1MIPS in decoder of 4kbps. The used memory is about program ROM 9.7kwords, table ROM 4.5kwords, RAM 5.1 kwords. The waveform of output is showed by the result of C simulator and Bit Exact. Also, for evaluation of speech quality of the vocoder of real-time implemented variable bit rate, it is estimated the MOS score of 3.69 in 4kbps.

• The Journal of the Acoustical Society of Korea
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• v.20 no.6
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• pp.34-39
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• 2001

An adaptive multi-rate (AMR) speech coder was adopted as a standard of W-CDMA by 3GPP and ETSI. The AMR coder is based on the CELP algorithm operating at rates ranging from 12.2 kbps down to 4.75 kbps, and it is a source controlled codec according to the channel error conditions and the traffic loading. In this paper, we implement the DSP S/W of the AMR coder using OakDSPCore. The implementation is based on the CSD17C00A chip developed by C&S Technology, and it is tested using test vectors, for the AMR speech codec, provided by ETSI for the bit exact implementation. The DSP B/W requires 20.6 MIPS for the encoder and 2.7 MIPS for the decoder. Memories required by the Am coder were 21.97 kwords, 6.64 kwords and 15.1 kwords for code, data sections and data ROM, respectively. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.