• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.127-134
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• 2005

Lost packet robustness is an most important quality measure for voice over IP networks(VoIP). Recovery of the lost packet from the received information is crucial to realize this robustness. So, this paper proposes the lost packet recovery method from the received information for real-time communication for CELP coder. The proposed BS-PLC (Both Side Packet Loss Concealment) based WSOLA(Waveform Shift OverLab Add) allow the lost packet to be recovered from both the 'previous' and 'next' good packet as the LP parameter and the excitation signal are respectively recovered. The burst of packet loss is modeled by Gilbert model. The proposed scheme is applied to G.729 most used in VoIP and is evaluated through the SNR(signal to noise) and the MOS(Mean Opinion Score) test. As a simulation result, The proposed scheme provide 0.3 higher in Mean Opinion Score and 2 dB higher in terms of SNR than an error concealment procedure in the decoder of G.729 at $20\%$ average packet loss rate.

• The Journal of the Acoustical Society of Korea
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• v.14 no.4
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• pp.38-45
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• 1995

In this paper a way to improve the performance of the long term prediction is proposed, which adopts the Multi-band Excitation (MBE) method in addition to the Code-Excited Linear Prediction (CELP) method at low bit rates below 4.8 kbps. In the proposed method, the multiband long term prediction is performed on the periodic components which still remain after the long term prediction of the conventional CELP method. At this point, the whole frequency region is divided into subbands whose size is equal to the spacing between the harmonics of the fundamental frequency, and the periodic multiband excitation signals. are represented as the sum of sine waves approximately as large as the spectrum of the excitation signals, so that the actual characteristics of the excitation signals can be better taken into account. To evaluate the performance of the proposed method, computer simulation is performed at 4.8 kbps. The 4.8 kbps DoD CELP and the 4.4 kbps IMBE were chosen as the reference vocoders for the speech quality measure. The result of the perceptual speech quality measure showed that the performance of the proposed method is better than that of the 4.8 kbps DoD CELP vocoder, and similar to that of the 4.4 kbps IMBE vocoder.

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.531-542
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• 2002

Novel tandemless transcoding method for AMR and EVRC speech coders is proposed in this paper. In contrast to conventional tandem method, the parameters which is used commonly in speech coder where CELP algorithm is adapted are directly transcoded. The proposed algorithm is composed of LSP transcoding, pitch delay transcoding, gains transcoding and fixed codebook vector transcoding Evaluation results show that the novel algorithm achieves better speech quality than tandem method and reduce computational complexity and delay.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12A
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• pp.2027-2035
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• 2001

In this paper, we present an audio stream delivery using the AMR (Adaptive Multi-Rate) coder that was adopted by ETSI and 3GPP as a standard vocoder for next generation IMT-2000 service in which includes combined sender (FEC) and receiver reconstruction technique in the Internet. By use of the media-specific FEC scheme, the possibility to recover lost packets can be much increased due to the addition of repair data to a main data stream, by which the contents of lost packets can be recovered. The AMR codec is based on the code-excited linear predictive (CELP) coding model. So we use a frame erasure concealment for CELP-based coders. The proposed scheme is evaluated with ITU-T G.729 (CS-ACELP) coder and AMR - 12.2 kbit/s through the SNR (Signal to Noise Ratio) and the MOS (Mean Opinion Score) test. The proposed scheme provides 1.1 higher in Mean Opinion Score value and 5.61 dB higher than AMR - 12.2 kbit/s in terms of SNR in 10% packet loss, and maintains the communicab1e quality speech at frame erasure rates lop to 20%.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1211-1214
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• 1998

Real-time implementation of Conjugate-Structure Algebraic CELP(CS-ACELP) is presented. ITU-T Study Group(SG) 15 has standardized the CS-ACELP speech coding algorithm as G.729. A real-time implementation of the CS-ACELP is achieved using 16 bit fixed point DSP16210 Digital Signal Processor (DSP) of Lucent Technologies. The speech coder has been implemented in the bit-exact manner using the fixed point CS-ACELP C source which is the part of the G.729 standard. To provide a multi-channel vocoder solution to digital communication system, we try to minimize the complexity(e.g., MIPS, ROM, RAM) of CS-ACELP. Our speech coder shows 15.5 MIPS in performance which enables 4 channel CS-ACELP to be processed with one DSP16210.

• The Journal of the Acoustical Society of Korea
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• v.27 no.1E
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• pp.11-16
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• 2008

In this paper, a pitch cycle waveform (PCW) is extracted, quantized, and interpolated in a time domain to synthesize high-quality speech at low bit rates. The pre-alignment technique is proposed for the accurate and efficient PCW extraction, which predicts the current PCW position from the previous PCW position assuming that pitch periods evolve slowly. Since the pitch periods are different frame by frame, the original PCW is converted into the fixed-dimension PCW using the dimension-conversion method, and subsequently quantized by code-excited linear predictive (CELP) coding. The excitation signal for the linear predictive coding (LPC) synthesis filter is generated using the time-domain interpolation and interlink of the quantized PCW's. The coder operates at 4.2 kbit/s and 3.2 kbit/s depending on the pitch period. Informal listening test demonstrates the effectiveness of the proposed coding scheme.

• Proceedings of the IEEK Conference
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• 2002.06a
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• pp.343-346
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• 2002

This paper presents a Prototype system developed for a Bluetooth interfaced VoIP system. The VoIP phone is developed based on tile implementation of a CELP coder on the TI 16bit DSP Processor A PC interfaced with Bluetooth module is used to designing a access point system. Host controller protocol stack is implemented to realize gateway between the wireless and wired line networks. A server application program for user management and call processing, which is based on TCP/IP peer to peer connection, is implemented for tile evaluation of overall interface system.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.9C
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• pp.877-886
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• 2005

Various frame recovery algorithms have been suggested to overcome the communication quality degradation problem due to Internet-typical impairments on Voice over IP(VoIP) communications. In this paper, we propose a new receiver-based recovery method which is able to enhance recovered speech quality with almost free computational cost and without an additional increment of delay and bandwidth consumption. Most conventional recovery algorithms try to recover the lost or erroneous speech frames by reconstructing missing coefficients or speech signal during speech decoding process. Thus they eventually need to modify the decoder software. The proposed frame recovery algorithm tries to reconstruct the missing frame itself, and does not require the computational burden of modifying the decoder. In the proposed scheme, the Vector Quantization(VQ) codebook indices of the erased frame are directly estimated by referring the pre-computed VQ Codebook Index Interpolation Tables(VCIIT) using the VQ indices from the adjacent(previous and next) frames. We applied the proposed scheme to the ITU-T G.723.1 speech coder and found that it improved reconstructed speech quality and outperforms conventional G.723.1 loss recovery algorithm. Moreover, the suggested simple scheme can be easily applicable to practical VoIP systems because it requires a very small amount of additional computational cost and memory space.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.12C
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• pp.1660-1668
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• 2004

This paper proposes a Voice Activity Detection (VAD) algorithm based on Radial Basis Function (RBF) network using dual threshold. The k-means clustering and Least Mean Square (LMS) algorithm are used to upade the RBF network to the underlying speech condition. The inputs for RBF are the three parameters in a Code Exited Linear Prediction (CELP) coder, which works stably under various background noise levels. Dual hangover threshold applies in BRF-VAD for reducing error, because threshold value has trade off effect in VAD decision. The experimental result show that the proposed VAD algorithm achieves better performance than G.729 Annex B at any noise level.

• The Journal of the Acoustical Society of Korea
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• v.20 no.6
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• pp.34-39
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• 2001

An adaptive multi-rate (AMR) speech coder was adopted as a standard of W-CDMA by 3GPP and ETSI. The AMR coder is based on the CELP algorithm operating at rates ranging from 12.2 kbps down to 4.75 kbps, and it is a source controlled codec according to the channel error conditions and the traffic loading. In this paper, we implement the DSP S/W of the AMR coder using OakDSPCore. The implementation is based on the CSD17C00A chip developed by C&S Technology, and it is tested using test vectors, for the AMR speech codec, provided by ETSI for the bit exact implementation. The DSP B/W requires 20.6 MIPS for the encoder and 2.7 MIPS for the decoder. Memories required by the Am coder were 21.97 kwords, 6.64 kwords and 15.1 kwords for code, data sections and data ROM, respectively. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.