• Journal of the Korean Institute of Telematics and Electronics
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• v.15 no.6
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• pp.1-7
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• 1978

This paper deals with speech digitization and bandwidth compression techniques, particularly two predictive coding methods-namely, adaptive differential pulse code modulation(ADPCM) and adaptive delta modulation(ADM). The principle of a typical adaptive quantizer that is used in ADPCM is explained, and discussed. Also, three companding methods(instantaueous, syllabic, and hybrid companding) that are used in ADM are explained in detail, and their performances are compared. In addition, the performances of ADPCM and ADM as speech coders are compared, and the inerits of each coder are discussed.

• Journal of the Korean Institute of Telematics and Electronics
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• v.15 no.3
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• pp.1-6
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• 1978

This paper deals with speech digitization and bandwidth compression techniques, particularly two predictive coding methods-namely, adaptive diferentia1 pulse code modulation(ADPCM) and adaptive delta modulation (ADM). The principle of a typical adoptive quantizer that is used in ADPCM is explained, and two analysis methods for the adaptive predictor coefficents, block and sequential analyses, are discussed. Also, three companding methods (instantaneous, syllabic, and hybrid companding) that are used in ADM are explained in detail, and their performances are compared. In addition, the performances of ADPCM and ADM as speech coders are compared, and the merits of each coder are discussed.

• Journal of Broadcast Engineering
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• v.11 no.2 s.31
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• pp.200-206
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• 2006

This paper presents a preprocessing method to modify the input audio signals of a speech coder to obtain the finally enhanced signals at the decoder. For the purpose, we introduce the noise suppression (NS) scheme and the adaptive gain control (AGC) where an audio input and its coding error are considered as a noisy signal and a noise, respectively. The coding error is suppressed from the input and then the suppressed input is level aligned to the original input by the following AGC operation. Consequently, this preprocessing method makes the spectral energy of the music input redistributed all over the spectral domain so that the preprocessed music can be coded more effectively by the following coder. As an artifact, this procedure needs an additional encoding pass to calculate the coding error. However, it provides a generalized formulation applicable to a lot of existing speech coders. By preference listening tests, it was indicated that the proposed approach produces significant enhancements in the perceived music qualities.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.4
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• pp.286-291
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• 2003

In this Paper, we designed an LSP(Line Spectral Pairs) parameter quantizer with cascaded structure of vector quantizer and scalar quantizer for the wideband speech coder. We have chosen the 16th-order of the LP coefficients. These coefficients are then transformed into the LSP parameters which have the excellent properties for quantization and easy stability checking condition of synthesis filter. In the first stage of quantization, input LSP parameters are split-vector-quantized using two 8-th order codebooks. In the second stage, the components of residual vector are individually quantized by the scalar quantizer utilizing the ordering property of LSP parameters. The designed adaptive VQ-SQ quantizer using 35 bits/frame shows the wideband transparency that the average spectral distortion should be less than 1.6 ㏈ and less than 4% of the frames should have SD above 3 ㏈. The simulation results show that the designed quantizer provides a 2-3 bits/frame saving over the typical vector-scalar quantizer.

• The Journal of the Acoustical Society of Korea
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• v.37 no.2
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• pp.110-117
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• 2018

In this paper, we propose a low delay window switching MDCT (Modified Discrete Cosine Transform) method for speech/audio coder. The window switching algorithm is used to reduce the degradation of sound quality in non-stationary trasient duration and to reduce the algorithm delay by using the low delay TDAC (Time Domain Aliasing Cancellation). While the conventional window switching algorithms uses overlap-add with different lengths, the proposed method uses the fixed overlap add length. It results the reduction of algorithm delay by half and 1 bit reduction in frame indication information by using 2 window types. We apply the proposed algorithm to G.729.1 based on MDCT in order to evaluate the performance. The propose method shows the reduction of algorithm delay by half while speech quality of the proposed method maintains same as the conventional method.

• MALSORI
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• no.57
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• pp.175-189
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• 2006

Because a packet loss brings about degradation of speech quality, VoIP speech coders have PLC (Packet Loss Concealment) mechanism. G.711, which is a mandatory VoIP speech coder, also has the PLC algorithm based on pitch period replication. However, it is not robust to burst losses. Thus, we propose two methods to improve the performance of the original PLC algorithm in G.711. One adaptively utilizes voiced/unvoiced information of adjacent good frames regarding to the current lost frame. The other is based on adaptive gain control according to energy variation across the frames. We evaluate the performance of the proposed PLC algorithm by measuring a PESQ value under different random and burst packet loss simulating conditions. It is shown from the experiments that the performance of the proposed PLC algorithm outperforms that of PLC employed in ITU-T Recommendation G.711.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.3
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• pp.119-124
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• 2009

In this paper, AMR-WB+ audio coder is implemented, in real-time, using Intel 400MHz Xscale PXA250 with 32bit RISC processor ARM9E-J(R)core. The assembly code for ARM9E-J(R)core is developed through the serial process of C code optimization, cross compile, assembly code manual optimization and adjusting the optimized code to Embedded Visual C++ platform. C code is trimmed on Visual C++ platform. Cross compile and assembly code manual optimization are performed on CodeWarrior with ARM compiler. Through these stages the code for both ARM EVM board and PDA is implemented. The average complexities of the code are 160.75MHz on encoder and 33.05MHz on decoder. In case of static link library(SLL), the required memories are 65.21Kbyte, 32.01Kbyte and 279.81Kbyte on encoder, decoder and common sources, respectively. The implemented coder is evaluated using 16 test vectors given by 3GPP to verify the bit-exactness of the coder.

• The Journal of the Acoustical Society of Korea
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• v.29 no.3E
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• pp.131-139
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• 2010

This paper proposes a novel spectral hole substitution technique for low bit-rate audio coding. The spectral holes frequently occurring in relatively weak energy bands due to zero bit quantization result in severe quality degradation, especially for harmonic signals such as speech vowels. The enhanced aacPlus (EAAC) audio codec artificially adjusts the minimum signal-to-mask ratio (SMR) to reduce the number of spectral holes, but it still produces noisy sound. The proposed method selectively predicts the spectral shapes of hole bands using either intra-band correlation, i.e. harmonically related coefficients nearby or inter-band correlation, i.e. previous frames. For the bands that have low prediction gain, only the energy term is quantized and spectral shapes are replaced by pseudo random values in the decoding stage. To minimize perceptual distortion caused by spectral mismatching, the criterion of the just noticeable level difference (JNLD) and spectral similarity between original and predicted shapes are adopted for quantizing the energy term. Simulation results show that the proposed method implemented into the EAAC baseline coder significantly improves speech quality at low bit-rates while keeping equivalent quality for mixed and music contents.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.289-292
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• 1998

본 논문에서는 CELP 음성 부호화기의 성능을 향상시키는 방법을 제안한다. 제안된 방법은 최적 코드북 검색 과정에서 추가적인 알고리듬의 지연 없이 미래 정보를 이용하고 두 인접한 코드북 부프레임 사이의 동시 최적화를 통하여 음성 부호화기의 성능을 향상시킨다. 또한, 제안된 코드북 검색 과정의 계산량을 조절하기 위한 방법도 제공된다. 제안된 방법의 성능을 검증하기 위하여 IS-96A QCELP 음성 부호화기를 이용하여 합성음의 스펙트럼과 Segmental SNR로 성능을 측정하는 모의실험을 실시하였으며, 제안된 방법을 적용한 QCELP 음성 부호화기가 기존의 QCELP에 비하여 향상된 성능을 보여주었다.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.131-135
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• 1998

5.6kbps 의 전송율에서 fixed codebook 으로 ISPP의 dynamic sparse algebraic codebook을 이용한 ACELP 알고리즘을 제안한다. 저전송율에서 음질에 중대한 영향을 미치는 대수적 방식의 고정코드북이 가지는 문제점을 최소화하여 음질의 증진을 꾀하였다. 또한 추가 계산량이 필요없는 U/V 분리기를 도입하여 LSF 보간시 발생하는 천이구간에서의 지연을 최소화하였다. 구현된 5.6 kbps ACELP 는 전화선상의 음질을 시료로 하여 주관적 음질면에서 6.3 kbps MP-MLQ와 동등하였으며 MNRU 15dB에서 약간 낮았다.