• Proceedings of the KSPS conference
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• 2004.05a
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• pp.235-238
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• 2004

Frame erasures cause speech quality degradation in wireless communication networks or packet networks. The degradation becomes worse when consecutive frame erasures occur. Speech coders have a frame erasure concealment(FEC) mechanism to compensate for frame erasures. It is meaningful to evaluate the performance of FEC mechanisms for frame erasures that occur in communications networks. In this paper, various frame erasures are designed. And the FEC algorithms of speech coders are evaluated and analyzed with the Perceptual Evaluation of Speech Quality(PESQ). It is found that the performances vary in accordance with frame erasure types, frame erasure rates, and utterance lengths.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.9 no.5
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• pp.129-132
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• 2009

Frame erasure is one of the most difficult problems in voice over IP (VoIP) networks and is a major source of speech quality degradation. In this paper, a frame erasure concealment algorithm based on nonlinear regression analysis is presented to minimize speech quality deterioration in code-excited linear prediction (CELP) based coders. We applied the proposed scheme to the ITU-T G.729 standard and obtained improved perceptual evaluation of speech quality (PESQ) scores compared to the conventional methods.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.4 no.1
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• pp.62-77
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• 2010

Emerging applications with high data rates will need to transport bulk data reliably in wireless sensor networks. ARQ (Automatic Repeat request) or Forward Error Correction (FEC) code schemes can be used to provide reliable transmission in a sensor network. However, the naive ARQ approach drops the whole frame, even though there is a bit error in the frame and the FEC at the bit level scheme may require a highly complex method to adjust the amount of FEC redundancy. We propose a bulk data transmission scheme based on erasure-resilient code in this paper to overcome these inefficiencies. The sender fragments bulk data into many small blocks, encodes the blocks with LT codes and packages several such blocks into a frame. The receiver only drops the corrupted blocks (compared to the entire frame) and the original data can be reconstructed if sufficient error-free blocks are received. An incidental benefit is that the frame error rate (FER) becomes irrelevant to frame size (error recovery). A frame can therefore be sufficiently large to provide high utilization of the wireless channel bandwidth without sacrificing the effectiveness of error recovery. The scheme has been implemented as a new data link layer in TinyOS, and evaluated through experiments in a testbed of Zigbex motes. Results show single hop transmission throughput can be improved by at least 20% under typical wireless channel conditions. It also reduces the transmission time of a reasonable range of size files by more than 30%, compared to a frame ARQ scheme. The total number of bytes sent by all nodes in the multi-hop communication is reduced by more than 60% compared to the frame ARQ scheme.

• 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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.10
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• pp.2227-2234
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• 2009

DVB-SSP is a new broadcasting system for hybrid satellite communications, which supports mobile handheld systems and fixed terrestrial systems. An upper layer, including erasure Reed-Solomon error correction combined with cyclic redundancy check. However, a critical factor must be considered in upper layer decoding. If there is only one bit error in an IP packet, the entire IP packet is considered as unreliable bytes, even if it contains correct bytes. If, for example, there is one real byte error, in an IP packet of 512 bytes, 511 correct bytes are erased from the frame. Therefore, this paper proposed upper layer decoding methods; LLR-based decoding. By means of simulation we show that the performance of the proposed decoding algorithm is superior to that of the conventional one.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.2A
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• pp.172-178
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• 2011

In this paper, the performance of bit interleaved of coded and modulation(BICM) module of the second generation of digital terrestrial television broadcasting system(DVB-T2) is evaluated with the help of computer simulation. The frame error rate performance is studied in AWGN, Rayleigh fading and 15% erasure channels. In addition, iterative receiver is considered that exchanges extrinsic information between the rotated demapper and the LDPC decoder. Through the simulation it is observed that under the flat fading Rayleigh channel, about 1.2dB gain at FER of $10^{-4}$ is introduced when rotated constellation and iterative demapping and decoding are employed. Under the 15% earasure channel, rotated constellation gives performance gain of about 5dB at BER of $10^{-4}$ and when IDD is applied, additional performance gain of about 3dB can be achieved.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.9C
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• pp.848-854
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• 2009

DVB-SSP is a new broadcasting system for hybrid satellite communications, which supports mobile handhold systems and fixed terrestrial systems. An upper layer, including erasure Reed-Solomon error correction combined with cyclic redundancy check. However, a critical factor must be considered in upper layer decoding. If there is only one bit error in an IP packet, the entire IP packet is considered as unreliable bytes, even if it contains correct bytes. If, for example, there is one real byte error, in an If packet of 512 bytes, 511 correct bytes are erased from the frame. Therefore, this paper proposed upper layer decoding methods; hybrid decoding. By means of simulation we show that the performance of the proposed decoding algorithm is superior to that of the conventional one in AWGN channel and TI channel.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.3
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• pp.19-28
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• 1998

This paper is concerned with the implementation of the equalization technique in a DCS 1800 system employing the soft-decision output Viterbi algorithm (SOVA), which makes the hardware complexity comparable to the hard decision MLSE and gives reliable performance. Also, the channel estimation technique with enhances the perfdormance of the soft-decision output equalizer is proposed, and the Viterbi decoder which operates effectively with the soft-decision output of the qualizer is implemented using the Very High Speed ICs Hardware Description Language (VHDL). From the simulation results, it is shown that the implemented Viterbi decoder operates effectively and the SOVA outperforms the hard-decision MLSE in terms of the frame erasure rate (FER) and bit error rate (BER).

• 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%.