• ETRI Journal
• /
• v.35 no.6
• /
• pp.990-1000
• /
• 2013

This paper describes the scalable extension of High Efficiency Video Coding (HEVC) to provide flexible high-quality digital video content services. The proposed scalable codec is designed on multi-loop decoding architecture to support inter-layer sample prediction and inter-layer motion parameter prediction. Inter-layer sample prediction is enabled by inserting the reconstructed picture of the reference layer (RL) into the decoded picture buffer of the enhancement layer (EL). To reduce the motion parameter redundancies between layers, the motion parameter of the RL is used as one of the candidates in merge mode and motion vector prediction in the EL. The proposed scalable extension can support scalabilities with minimum changes to the HEVC and provide average Bj${\o}$ntegaard delta bitrate gains of about 24% for spatial scalability and of about 21% for SNR scalability compared to simulcast coding with HEVC.

• Journal of the Korean Institute of Telematics and Electronics A
• /
• v.33A no.5
• /
• pp.172-179
• /
• 1996

In the paper, QCELP (qualcomm code excited linear predictive), CDMA (code division multiple access)'s vocoder algorithm, was analyzed. And then, a ptich parameter seaarch architecture for 16-bit programmable DSP(digital signal processor) for QCELP was designed. Because we speed up the parameter search through high speed DSP using two MACs, we can satisfy speech codec specifiction for the digital celluar. Also, we implemented in FIFO(first-in first-out) memory using register file to increase the access time of data. This DSP was designed using COMPASS, ASIC design tool, by top-down design methodology. Therefore, it is possible to cope with rapid change at mobile communication market.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.21 no.6
• /
• pp.1370-1378
• /
• 1996

A subjective video quality assessment methods are proposed based on CCIR Rec. 500-5 for an evaluation and testing of compressed video quality and performance of video codec to be designed in accordance with the MPEG-2 MP ML specification which is adapted as a DTV standard for Korea digital DBS. Video sequence compressed in compliance with MPEG-2 MP ML encoding parameterswastested by the proposed video quality evaluation procedure. Test sequence were compressed at the bit rate 6Mbps, 7.5Mbps and 9Mbps, repectively. Test results of the 7.5Mbps bitrate showed a satisfactory picture quality at about 4.0 on the 5.0 absolute scale of ITU-R 500-5.

• Proceedings of the Acoustical Society of Korea Conference
• /
• 1994.06a
• /
• pp.1070-1075
• /
• 1994

This paper describes the method for designing loudness ratings as transmission quality for ISDN telephone connected to fully digital network. To design the desirable loudness ratings for ISDN telephone, the model system of digital speech communication for subjective test is developed and opinion tests for establishing the optimal CODEC input level, the range of overall loudness rating, and sidetone masking rating are performed. As the results, the desirable ranges of loudness ratings are proposed as 6 to 8dB for sending, 0 to 2dB for receiving, and 10 to 14dB for sidetone masking rating.

• ETRI Journal
• /
• v.31 no.6
• /
• pp.732-740
• /
• 2009

This paper describes the implementation of a digital audio effect system-on-a-chip (SoC), which integrates an embedded digital signal processor (DSP) core, audio codec intellectual property, a number of peripheral blocks, and various audio effect algorithms. The audio effect SoC is developed using a software and hardware co-design method. In the design of the SoC, the embedded DSP and some dedicated hardware blocks are developed as a hardware design, while the audio effect algorithms are realized using a software centric method. Most of the audio effect algorithms are implemented using a C code with primitive functions that run on the embedded DSP, while the equalization effect, which requires a large amount of computation, is implemented using a dedicated hardware block with high flexibility. For the optimized implementation of audio effects, we exploit the primitive functions of the embedded DSP compiler, which is a very efficient way to reduce the code size and computation. The audio effect SoC was fabricated using a 0.18 ${\mu}m$ CMOS process and evaluated successfully on a real-time test board.

• The Journal of the Acoustical Society of Korea
• /
• v.24 no.7
• /
• pp.416-422
• /
• 2005

Speech recognition in digital communication systems has very low performance due to the spectral distortion caused by speech codecs. In this paper, the spectral distortion by speech codecs is analyzed and a pre-processing method which compensates for the spectral distortion is proposed for performance enhancement of speech recognition. Three standard speech codecs. IS-127 EVRC. ITU G.729 CS-ACELP and IS-96 QCELP. are considered for algorithm development and evaluation, and a single method which can be applied commonly to all codecs is developed. The performance of the proposed method is evaluated for three codecs, and by using the speech features extracted from the compensated spectrum. the recognition rate is improved by the maximum of $15.6\%$ compared with that using the degraded speech features.

• Proceedings of the Korean Information Science Society Conference
• /
• 2005.11b
• /
• pp.607-609
• /
• 2005

본 논문에서는 실시간으로 다채널의 카메라에서 입력되는 영상을 동시에 압축 및 복원할 수 있는 PC 기반의 DVR (Digital Video Recording) 시스템을 위한 MPEG-4 방식의 코덱을 구현하고자 한다. 현재까지의 일반적인 압축 방식은 화상회의 용도의 H.263, VCD 화질의 MPEG-1, DVD급 화질인 MPEG-2가 널리 적용되고 있다. 하지만 이러한 방법들은 저장 데이터의 양이 커서 효율적인 저장이 어렵다. 따라서 본 논문에서는 이러한 문제점을 해결하면서 실시간적으로 다채널 영상 데이터 저장이 가능한 MPEG-4 압축 방식을 적용한 코덱을 제작한다.

• Journal of the Korean Institute of Telematics and Electronics
• /
• v.27 no.4
• /
• pp.491-501
• /
• 1990

Recently, four speech coding techniques, namely, SBC-APCM(sub-band coding adaptive PCM), RPE-LPC(regualr pulse excitation linear predictive codec), MPE-LTP(multi-pulse excited long-term prediction) and CELP (code-excited linear prediction) are proposed for digital mobile radio applications. However, a performance comparison of these coders in the Rayleigh fading environment has not been made yet. In this paper, the performances of the four spech coders in the random bit error and burst error environment are investigated. For the channel coding of SBC-APCM, RPE-LPC and MPE-LTP, the sensitivity of output bit stream is measured and a bit selective forward error correction is provided acording to the measured bit sensitivity. And for an attempt to improve the performance of CELP, an optimum quantizer is applied for transmitting scalar quantities in CELP. However, an improvement over the conventional approach is found to be negligible. For the channel coding of CELP, Reed-Solomon code, Golay code, convolutional code of rate 1/2 shows the best performance. Finally, from the simulation results, it is concluded that CELP is the best candidate for digital mobile radio and is followed by MPE-LTP, SBC-APCM and RPE-LPC.

• Journal of Practical Engineering Education
• /
• v.6 no.2
• /
• pp.91-97
• /
• 2014

Education equipment for field programmable gate array (FPGA) based multimedia player design is introduced. Using the education equipment, an example of hardware design for color detection and augment reality (AR) game is described, and an example of syllabus for "Digital system design using FPGA" course is introduced. Using the education equipment, students can develop the ability to design some hardware, and to train the ability for the creative capstone design through conceptual, partial-level, and detail designs. By controlling audio codec, system-on-chip (SOC) design skills combining a NIOS II soft microprocessor and digital hardware in one FPGA chip are improved. The ability to apply wireless communication and LabView to FPGA-based digital design is also increased.

• Journal of the Korea Society of Computer and Information
• /
• v.13 no.3
• /
• pp.99-106
• /
• 2008

Recently, digital music retrieval is using in many fields (Web portal. audio service site etc). In existing fields, Meta data of music are used for digital music retrieval. If Meta data are not right or do not exist, it is hard to get high accurate retrieval result. Contents based information retrieval that use music itself are researched for solving upper problem. In this paper, we propose Same music recognition method using similarity measurement. Feature data of digital music are extracted from waveform of music using Simplified MFCC (Mel Frequency Cepstral Coefficient). Similarity between digital music files are measured using DTW (Dynamic time Warping) that are used in Vision and Speech recognition fields. We success all of 500 times experiment in randomly collected 1000 songs from same genre for preying of proposed same music recognition method. 500 digital music were made by mixing different compressing codec and bit-rate from 60 digital audios. We ploved that similarity measurement using DTW can recognize same music.