• The Journal of the Acoustical Society of Korea
• /
• v.19 no.3
• /
• pp.26-34
• /
• 2000

In this paper, we propose a 4kbps speech coder that combines the harmonic vector excitation coding with time-separated transition coding. The harmonic vector excitation coding uses the harmonic excitation coding in the voiced frame and uses the vector excitation coding with the structure of analysis-by-synthesis in the unvoiced frame, respectively. But two mode coding method is not effective for transition frame mixed in voiced and unvoiced signal and a new method beyond using unvoiced/voiced mode coding is needed. Thus, we designed a time-separated transition coding method for transition frame in which a voiced/unvoiced decision algorithm separates unvoiced and voiced duration in a frame, and harmonic-harmonic excitation coding and vector-harmonic excitation coding method is selectively used depending on the previous frame U/V decision. In the decoder, the voiced excitation signals are generated efficiently through the inverse FFT of harmonic magnitudes and the unvoiced excitation signals are made by the inverse vector quantization. The reconstructed speech signal are synthesized by the Overlap/Add method.

• The Journal of the Acoustical Society of Korea
• /
• v.22 no.3E
• /
• pp.101-109
• /
• 2003

A multi-mode harmonic transform coding (MMHTC) for speech and music signals is proposed. Its structure is organized as a linear prediction model with an input of harmonic and transform-based excitation. The proposed coder also utilizes harmonic prediction and an improved quantizer of excitation signal. To efficiently quantize the excitation of music signals, the modulated lapped transform(MLT) is introduced. In other words, the coder combines both the time domain (linear prediction) and the frequency domain technique to achieve the best perceptual quality. The proposed coder showed better speech quality than that of the 8 kbps QCELP coder at a bit-rate of 4 kbps.

• The Journal of the Acoustical Society of Korea
• /
• v.23 no.5
• /
• pp.420-427
• /
• 2004

This paper presents real time implementation of 4kbps EHSX (Enhanced Harmonic Stochastic Excitation) speech coder that combines the harmonic vector excitation coding with time-separated transition coding. The harmonic vector excitation coding uses the harmonic excitation coding for voiced frames and used the vector excitation coding with the structure of analysis-by-synthesis for unvoiced frames, respectively. For transition frames mixed with voiced and unvoiced signal, we use the time-separated transition coding. In this paper. we present the optimization methods of implementation speech coder on the EMS320C6701/sup (R)/ DSP. To reduce the complex for real-time implementation. we perform the optimization method in algorithm by replacing the complex sinusoidal synthesis method with IFFT. and we apply fully pipelines hand assembly coding after converting it from floating source to fixed source. To generate a more efficient code. we also make use or the available EMS320C6701/sup (R)/ resources such as Fastest67x library and memory organization.

• The Journal of the Acoustical Society of Korea
• /
• v.20 no.4E
• /
• pp.24-30
• /
• 2001

This paper describes a fast harmonic estimation, referred to as Delta Adjustment (DA), using a low resolution pitch. The presented DA method is based on modification of the Generalized Dual Excitation (GDE) technique[1] which was proposed to improve speech enhancement performance. We introduce the GDE technique and modify it to be suitable for low bit rate harmonic coding that uses only an integer pitch estimate. Unlike the GDE, the DA matches a frequency-warped version of the original spectrum that conforms to a fixed pitch at all harmonic bands. In addition, complexity and performance of the presented method are described in comparison with those of the conventional Fractional Pitch (FP) based harmonic estimation. Experimental results showed that the DA algorithm significantly reduces the complexity of the FP method while maintaining the performance.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.28 no.3C
• /
• pp.239-247
• /
• 2003

This paper presents a dual-rate (2.4/4.0 kbps) Improved Harmonic-CELP(IHC) speech coder based on the EHC(Efficient Harmonic-CELP) which was presented by the authors. The proposed IHC employs the harmonic coding for voiced and the CELP for unvoiced segments. In the IHC, an initial voiced/unvoiced estimate is obtained by the pitch gain and energy. Then, the final V/UV mode is decided by using the frame energy contour. A new harmonic estimation combining peak picking and delta adjustment provides a more reliable harmonic estimation than that in the EHC. In addition, a noise mixing scheme in conjunction with an improved band voicing measurement provides the naturalness of the synthesized speech. To demonstrate the performance of the proposed IHC coder, the coder has been implemented and compared with the 2.0/4.0 kbps HVXC(Harmonic excitation Vector Coding) standardized by MPEG-4. Results of subjective evaluation showed that the proposed IHC coder and produce better speech quality than the HVXC, with only 40% complexity of the HVXC.

• 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.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.42 no.2 s.302
• /
• pp.131-142
• /
• 2005

The low rate speech coders under 4 kbit/s are based on sinusoidal transform coding (STC) or multiband excitation (MBE). Since the harmonic coders are not efficient to reconstruct the transient segments of speech signals such as onsets, offsets, non-periodic signals, etc, the coders do not provide a natural speech quality. This paper proposes method of a efficient transient model :d a multi-mode low rate coder at 2.4 kbit/s that uses harmonic model for the voiced speech, stochastic model for the unvoiced speech and a model using aperiodic pulse location tracking (APPT) for the transient segments, respectively. The APPT utilizes the harmonic model. The proposed method uses different models depending on the characteristics of LPC residual signals. In addition, it can combine synthesized excitation in CELP coding at time domain with that in harmonic coding at frequency domain efficiently. The proposed coder shows a better speech quality than 2.4 kbit/s version of the mixed excitation linear prediction (MELP) coder that is a U.S. Federal Standard for speech coder.

• The Journal of the Acoustical Society of Korea
• /
• v.19 no.7
• /
• pp.53-58
• /
• 2000

본 논문은 주파수 전이신호와 시간 전이 신호에 대해서 고조파 잡음 여기 방법과 시간 분리 여기 방법을 적용한 2.4 kbps 음성부호화 방법을 제안한다. 혼합 여기 부호화 방법은 주기 신호와 비 주기 신호를 효과적으로 표현하기 위해 하모닉 잡음 모델을 사용한다. 혼합신호에 대한 잡음 성분은 캡스트럴 분석 방법을 사용함으로써 추출되고, AR (Autoregressive Model) 모델에 의해 표현된다. 시간 전이구간 신호에서의 모호한 음성을 효과적으로 제거하기 위한 또 다른 방법이 제안된다. 제안된 시간 분리 방법은 시간 에너지 변화정도를 관찰함으로써 전이 시점을 감지하고 다른 시간 길이를 가지는 두 블록으로 분리하여 분석한다. 시간 분리 방법은 분석을 위한 비대칭 윈도우와 합성에서의 위상 합성 방법을 포함한다. 제안된 방법을 사용한 2.4 kbps 음성부호화 방법은 주관적 음질 평가에서 전이구간에서의 지각적 음질의 향상을 보여주었으며, 원본 음성 스펙트럼과의 고조파 비 매칭에 의한 윙윙거리는 기계적인 잡음을 감소시킨다.

• Journal of Korea Multimedia Society
• /
• v.16 no.10
• /
• pp.1171-1179
• /
• 2013

Interactive audio service provide with audio generating and editing functionality according to user's preference. A spatial audio object coding (SAOC) scheme is audio coding technology that can support the interactive audio service with relatively low bit-rate. However, when the SAOC scheme remove the specific one object such as vocal object signal for Karaoke mode, the scheme support poor quality because the removed vocal object remain in the SAOC-decoded background music. Thus, we propose a new SAOC vocal harmonic extranction and elimination technique to improve the background music quality in the Karaoke service. Namely, utilizing the harmonic information of the vocal object, we removed the harmonics of the vocal object remaining in the background music. As harmonic parameters, we utilize the pitch, MVF(maximum voiced frequency), and harmonic amplitude. To evaluate the performance of the proposed scheme, we perform the objective and subjective evaluation. As our experimental results, we can confirm that the background music quality is improved by the proposed scheme comparing with the SAOC scheme.

• Proceedings of the IEEK Conference
• /
• 1998.10a
• /
• pp.1247-1250
• /
• 1998

In speech synthesis, LF model is widely used for excitation signal for voice source coding system. But LF model does not represent the harmonic frequencies of excitation signal. We propose an effective method which use sinusoidal functions for representing the harmonics of voice source signal. The proposed method could achieve more exact voice source waveform and better synthesized speech quality than LF model.