• The Journal of the Acoustical Society of Korea
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• v.20 no.8
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• pp.19-23
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• 2001

DSP-based implementation is pervasive in wireless communication parts for systems and handsets according to developing high-speed and low-power programmable Digital Signal Processor (DSP). In this paper, we present a real-time implementation of multi-channel Adaptive Multi-rate (AMR) speech coder. The real-time implementation of an AMR algorithm is achieved using 32-bit fixed-point TMS320C6202 DSP chip that operates at 250 MHz. We performed cross compile, linear assembly optimization and TMS320C62xx assembly optimization for real-time implementation. Furthermore, speech data input/output function and communication function with external CPU is included in an AMR speech coder. The AMR Speech coder developed using DSP EVM board was evaluated in ETRI IMT-2000 Test-bed system.

• The Journal of the Acoustical Society of Korea
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• v.17 no.3E
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• pp.53-57
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• 1998

Spectral information at a speech coder should be quantized with sufficient accuracy to keep perceptually transparent output speech. Spectral information at a low bit rate speech coder is usually transformed into corresponding line spectrum pair parameters and is often quantized with a vector quantization algorithm. As the vector quantization algorithm generally has high complexity in the optimal code vector searching routine, the complexity reduction in that routine is investigated using the ordering property of the line spectrum pair. When the proposed complexity reduction algorithm is applied to the well-known split vector quantization algorithm, the 46% complexity reduction is achieved in the distortion measure compu-tation.

• 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.

• Proceedings of the KIPE Conference
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• 2002.07a
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• pp.727-729
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• 2002

In this paper, we proposed quad variable rates ADPCM coding method and its implementation on C6000 DSP, which is modified from the standard ADPCM of ITU G.726 for speech quality improvement considering the environmental noise Four coding rates, 16Kbps, 24Kbps, 32Kbps and 40Kbps are used for speech window samples and the rate decision threshold is decided by the environmental noise level. The object of the proposed method is to reduce the coding rate while retaining the speech quality and the speech quality is considerably close to 40Kbps single rate coder with the coding rate close to 16Kbps single rate coder under the environmental noise. The environmental noise level affects the coding rate and the noise level is calculated per every speech window samples. At high noise level, more samples are coded at higher rates to enhance the quality, but at low noise level, only the big speech signals are coded at higher rates and more speech samples are coded at lower coding rates to reduce the coding rates. The influence of the noise on tile speech signal is considerably high for small signals and the small signal has the higher ZCR (zero crossing rate). The method is simulated in PC and to be implemented on C6000 floating point DSP board in real time operations.

• 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.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 $80.8\%$.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.16 no.1
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• pp.85-100
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• 1991

In the discussion of digital mobile communication systems the speech coder based on the high quality low bit rate is an essential part of topics to overcome the limited availability of radio spectrum, which will enhance the communication services. In this paper we present the implementation and performance evaluation of 13kbps RPE LTP speech coder. An implementation of a real time full duplex coder with 75% of DSP loading rate using a single DSP chip has been shown, and also the fixed point simulations for H/W implementation has been performed. Finally, analysis result for relative bit importance of each transmitting parameter has been shown for channel coding.

• Proceedings of the IEEK Conference
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• 2003.11a
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• pp.57-60
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• 2003

In this paper, we propose a new excitation enhancement technique to improve the speech quality of low bit rate speech 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 parameters estimation and harmonic generation. and apply the 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 IEEK Conference
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• 2001.06d
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• pp.115-118
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• 2001

This paper presents dispersed-pulse and random codebook for CELP coder. This coder operates on speech frames of 20ms and generates an excitation vector by convoluting dispersion vectors with signed pulses in an algebraic codevector. The improvement of pulse-based fixed codebook is performed at a low bit rate. A high performance fixed-codebook consists of a partial algebraic codebook and a random codebook in unvoiced and stationary noise regions. The proposed CELP coder is quantized with 4kb/s and is compared with G.729 (Bkb/s CS-ACELP). Subjective testing shows better quality than reference coders under some background noise conditions

• The Journal of the Acoustical Society of Korea
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• v.31 no.7
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• pp.472-479
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• 2012

This paper is aimed to design 9.2 kbps low bits late transform excitation coder that target to voice and audio signal. To set up low bit rate, we used Band-selection in frequency domain and gain-shape quantization and AbS structure. To decrease lots of calculation from ABS structure, we used each band IDFT and synthesis. And we designed non-transfer band for performance by inserting comfort noise. We propose coder that has low bit rate and similar performance comparing with original 10.4 kbps AMR-WB+ TCX mode.