• 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.

• The Journal of the Acoustical Society of Korea
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• v.22 no.3E
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• pp.101-109
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• 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 Korean Institute of Communications and Information Sciences
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• v.28 no.3C
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• pp.239-247
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• 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
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• v.18 no.12
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• pp.1806-1814
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• 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 the Acoustical Society of Korea
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• v.23 no.5
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• pp.420-427
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• 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.

• Speech Sciences
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• v.11 no.2
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• pp.259-269
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• 2004

In this paper, we propose a new excitation enhancement technique to improve the speech quality of low bit-rate code-excited linear prediction (CELP) coders. The proposed technique is based on a harmonic model and it is employed only in the decoding process of speech coders without any additional bits. We develop the procedure of harmonic model parameter estimation and harmonic generation, and apply this technique to a current state-of-the-art low bit rate speech coder, ITU-T G.729 Annex D. Also, its performance is measured by using the ITU-T P.862 PESQ score and compared to those of the phase dispersion filter and the long-term postfilter applied to the decoded excitation. It is shown that the proposed excitation enhancement technique can improve the quality of decoded speech and provide better quality for male speech than other techniques.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.195-198
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• 2002

This paper presents an improved MELP Coder using Fourier magnitude compensation method chosen the new 2.4 kbit/s U.S. federal Standard. Although the MELP is quite good, it has some distortion for low-pitch male speakers. An improved MELP coder includes a post processing for the fourier magnitude model that allows the MELP to reconstruct the lower frequency spectrum more accurately and improve the speech quality. In this new compensation algorithm, the harmonic magnitudes in the low frequencies are adaptively modified by removing the effect of the two filters. Also, the bit rate of the improved MELP coder is the same as that of the Federal Standard MELP coder. formal quality tests show that the improved MELP coder was preferred over the Federal Standard MELP coder by $80.8\%$.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.2 s.302
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• pp.131-142
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• 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 Korean Institute of Communications and Information Sciences
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• v.29 no.7C
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• pp.962-970
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• 2004

This paper presents an efficient implementation of the 2.4 kbps EHSX(Enhanced Harmonic Stochastic Excitation) speech coder on a TMS320C6701$^{TM}$ floating-point digital signal processor. The EHSX speech codec is based on a harmonic and CELP(Code Excited Linear Prediction) modeling of the excitation signal respectively according to the frame characteristic such as a voiced speech and an unvoiced speech. In this paper, we represent the optimization methods to reduce the complexity for real-time implementation. The complexity in the filtering of a CELP algorithm that is the main part for the EHSX algorithm complexity can be reduced by converting program using floating-point variable to program using fixed-point variable. We also present the efficient optimization methods including the code allocation considering a DSP architecture and the low complexity algorithm of harmonic/pitch search in encoder part. Finally, we obtained the subjective quality of MOS 3.28 from speech quality test using the PESQ(perceptual evaluation of speech quality), ITU-T Recommendation P.862 and could get a goal of realtime operation of the EHSX codec.c.

• The Journal of the Acoustical Society of Korea
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• v.22 no.8
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• pp.717-726
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• 2003

This paper presents the methods that design the Wavelet-based pitch detector for 2,4 kbps Harmonic-CELP Coder, and that achieve the effective waveform interpolation by decision window shape of the transition region, Waveform interpolation coder operates by encoding one pitch-period-sized segment, a prototype segment, of speech for each frame, generate the smooth waveform interpolation between the prototype segments for voiced frame, But, harmonic synthesis of the prototype waveforms between previous frame and current frame occur not only waveform errors but also discontinuity at frame boundary on that case of pitch halving or doubling, In addtion, in transition region since waveform interpolation coder synthesizes the excitation waveform by using overlap-add with triangularity window, therefore, Harmonic-CELP fail to model the instantaneous increasing speech and synthesis waveform linearly increases, First of all, in order to detect the precise pitch period, we use the hybrid 1st pitch detector, and increse the precision by using 2nd ACF-pitch detector, Next, in order to modify excitation window, we detect the onset, offset of frame by GCI, As the result, pitch doubling is removed and pitch error rate is decreased 5.4% in comparison with ACF, and is decreased 2,66% in comparison with wavelet detector, MOS test improve 0.13 at transition region.