• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.18 no.12
• /
• pp.1806-1814
• /
• 1993

Efficient speech coders using tree coding combined with harmonic scaling are designed at the rate of 4.8 kilobitts/sec (kbps). A time domain harmonic scaling algorithm (TDHS) is used to compress input speech by a factor of two. This process allows the tree coder have 1.5 bits/sample for 4.8 kbps in the case of a 6.4 kHz sampling rate. In the backward adaptive tree coder, there are three components of the code generator, including a hybrid adaptive quantizer, a short-term predictor and a pitch predictor. The robustness of the tree coder is achieved by carefully choosing the input of the short term predictor adaptation. Also, inclusion of a smoother in the pitch predictor improves the error performance of tree coder in the noisy channel. Subjectively, tree coding combined with TDHS provides good quality speech at 4.8 kbps.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.23 no.7
• /
• pp.1650-1658
• /
• 1998

This paper describes how to remove echoes effectively using speech parameter information provided form speech coder. More specially, the proposed adaptive echo canceller utilizes the excitation signal or linearly predicted error signal instead of output speech signal of vocoder as the input signal for adaptation algorithm. The normalized least mean ssquare(NLMS) algorithm is used for the adaptive echo canceller. The proposed algorithm showed a fast convergece charactersitcis in the sinulatio compared to the conventional method. Specially, the proposed echo canceller utilizing the excitation signal of speech coder showed about four times fast convergence speed over the echo canceller utilizing the output speech signal of the speech coder for the adaptation input.

• The Journal of the Acoustical Society of Korea
• /
• v.3 no.2
• /
• pp.69-90
• /
• 1984

본 논문에서는 잡음 spectrum 변형 기능을 가진 APCM, ADPCM 및 ADM 음성 부호기의 성능 에 관해서 연구하였다. 잡은 SPECTRUM 변형방식은 두가지를 고려할 수 있는데, APCM과 ADPCM에 서는 C-massage weighting 된 양자화 잡음을 최소화하는 noise feedback filter를 이용하는 방법을 채택 하고, ADM에서는 in-band의 잡음의 일부를 신호대역의 밖으로 옮기는 방법을 사용하였다. APCM 과 ADPCM 부호기의 성능을 측정하는데는 주파수가 weighting이 된 신호대 잡음비와 segment된 FWSQNR를 사용하였다. 실제음성을 사용한 simulation 결과에 의하면 잡음 spectrum 변형기능을 가진 부호기가 없는 것보다 0.5 내지 3dB 가량 좋은 것으로 나타났다. 이러한 개선은 양적으로 비교적 적은 것이 사실이지만 실제로 음성을 들어보면 음질이 현저히 좋아짐을 알 수 있었다.

• The Journal of the Acoustical Society of Korea
• /
• v.21 no.5
• /
• pp.462-470
• /
• 2002

Wideband speech, characterized by a bandwidth of 50∼7000 ㎐, sounds more natural and intelligible, and is less tiring to listen to when compared to narrowband speech characterized by a bandwidth of 300∼3400 ㎐. Wideband speech coders, however, have not been as successful as the narrowband speech coders because of their higher bit rate. In this paper, we propose a new wideband speech coder which combines the European standard of a narrowband speech coder, i.e., GSM-EFR, and a transform coder using the discrete wavelet transform. The proposed wideband speech coder operates as follows input speech is first split into two subbands with equal bandwidth and the two subband signals are coded and decoded by each subband coder. A GSM-EFR is adopted as a lower subband coder and a subband coder with wavelet transformed speech is designed for a upper subband coder. The total bit rate of the proposed coder is 18.9kbps (12.2 kbps for lower band coder and 6.7 kbps for upper band coder), and informal listening test results have shown that the proposed coder has comparable speech quality to that of G.722 with 56 kbps.

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

• Proceedings of the Acoustical Society of Korea Conference
• /
• 1995.06a
• /
• pp.246-249
• /
• 1995

4.16kbps의 전송률을 갖는 음성 부호화기 구조에 관하여 기술한다. 제안된 음성 부호화기는 개방 회로 피치 검출기와 이로부터 생성된 pulse train을 코드북으로 갖는 CELP 부호화기이다. Pulse-Train codebook은 분석 프레임별로 부호화 및 복호화 양단에서 생성되며 음성의 피치 및 포만트 정보를 내포하고 있다. 구현된 PT-CELP는 random codebook 방식의 CELP에 비해 적은 크기로 codebook을 만들 수 있으며 음성의 특징을 충분히 반영하므로 합성된 음성의 음질을 향상시킬 수 있다.

• The Journal of the Acoustical Society of Korea
• /
• v.20 no.3
• /
• pp.34-41
• /
• 2001

In an IMBE (Improved Multi-band Excitation) speech coder, initial pitch estimation occupies most of the total computing time for the coder due to complex cost function and exhaustive search over candidate pitches. Future frames in initial pitch estimation cause inevitable time delay. Therefore, it is difficult to implement a real-time coder. Furthermore, unvoiced frames use the unnecessary pitch estimation as in the voiced frames. In this paper, each frame is determined voiced or unvoiced by Dyadic Wavelet Transform (DyWT) and, then, initial pitch estimation is performed only for voiced frame. Therefore different pitch estimation algorithms are employed between voiced and unvoiced frames incurring reduced time delay at transmitter and receiver. Simulation result show that the relative complexity of initial pitch estimation is reduced by 23％, and the processing time decreases down to 1/10 ∼ 1/1l of the IMBE coder while speech quality is almost maintained.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.18 no.3
• /
• pp.348-354
• /
• 1993

In this research, a multi-tree coder at 9.6Kbits/sec using a novel scheme for adaptation of the short-term coefficients is developed. The overall delay of the tree coder is maintained at 2.5 msec(16 samples at the 6.4KHz sampling frequency). This coder produces good quality speech over ideal channels, and it is very robust to channel errors up to a bit error rate (BER) of $10^{-3}$. This robustness is achieved by using a parallel adaptation scheme in combination with the use of a smoothed version of the received excitation sequence for adaptation of the short-term prediction coefficients. For the multi-tree coder, reconstructed output speech is evaluated using signal-to-quantization noise ratios (SNR), segmental SNRs, and informal listening tests.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.13 no.6
• /
• pp.55-61
• /
• 2013

In this paper, we describe design and implementation of audio encoder/decoder for voice synthesis. It uses the encoding of difference value of successive samples instead of the original sample value. and has the compression ratio of 4. The function is verified by using FPGA and the performance is measured by the fabricated chip using $0.35{\mu}m$ standard CMOS process. The system clock is 16.384MHz. The measured THD+n is from -40dB to -80dB with frequency variation and the power consumption is about 80mW. It is suited for the mobile application of high audio quality and low power consumption.

• Proceedings of the Acoustical Society of Korea Conference
• /
• 1998.08a
• /
• pp.465-469
• /
• 1998

가변 전송율 음성 부호화기를 위한 음성 검출기를 통계적 모델을 적용하여 설계한다. 제안된 음성 검출기는 음성 파라미터를 decision-directed 방식으로 추정함으로써 LRT를 이용하여 동작 특성이 우수한 판정 규칙을 유도한다. 또한 음성 발생 사건들을 1차의 Markov process 로 모델링 함으로써 과거의 관찰들을 현재 프레임의 음성 검출 과정에서 고려할 수 있는 행오버 알고리즘을 개발한다. 개발된 음성 검출기는 고려된 실험환경에서 ITU-T 표준인 G.729 Annex B 음성 검출기보다 맹 우수한 성능을 나타내었다.