• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.27 no.1A
• /
• pp.26-34
• /
• 2002

Mobile telecommunication systems such as IS-95 and IMT-2000 employ frame based communication using frames up to 20 msec in length and the receiving end has to store the whole frome before it is being processed. The size of the frame buffer ofter dominates those of the processing unit such as soft decision Viterbi decoder. The frame buffer for IMT-2000, for example, has to be increased 80 times as large as that of IS-95. One of the parameters deciding the number of bits in a frame will be obviously the number of bits in soft quantization. Start after striking space key 2 times. This paper has studied a new aspect of offset and quantization step size on BER performance and proposes a new 3-bit soft quantization algorithm which shows similar performance as that of 4-bit soft decision Viterbi receiver. The optimal offset values and step sizes for the other practical quantization levels ---16, 8, 4, 2--- have also been found. In addition, a new optimal symbol metric table has been devised which takes the accumulation value of various repeated signals and produces a rescaled 3-bit valu.tart after striking space key 2 times.

• Journal of the Korea Academia-Industrial cooperation Society
• /
• v.17 no.7
• /
• pp.49-53
• /
• 2016

Recently, the use of signal compression methods to improve the efficiency of wireless networks have increased. In particular, the MPC system was used in the pitch extraction method and the excitation source of voiced and unvoiced to reduce the bit rate. In general, the MPC system using an excitation source of voiced and unvoiced would result in a distortion of the synthesis speech waveform in the case of voiced and unvoiced consonants in a frame. This is caused by normalization of the synthesis speech waveform in the process of restoring the multi-pulses of the representation segment. This paper presents an ACFBD-MPC (Amplitude Compensation Frequency Band Division-Multi Pulse Coding) using amplitude compensation in a multi-pulses each pitch interval and specific frequency to reduce the distortion of the synthesis speech waveform. The experiments were performed with 16 sentences of male and female voices. The voice signal was A/D converted to 10kHz 12bit. In addition, the ACFBD-MPC system was realized and the SNR of the ACFBD-MPC estimated in the coding condition of 8kbps. As a result, the SNR of ACFBD-MPC was 13.6dB for the female voice and 14.2dB for the male voice. The ACFBD-MPC improved the male and female voice by 1 dB and 0.9 dB, respectively, compared to the traditional MPC. This method is expected to be used for cellular telephones and smartphones using the excitation source with a low bit rate.

• Journal of the Institute of Electronics and Information Engineers
• /
• v.53 no.12
• /
• pp.141-146
• /
• 2016

HEVC/H.265 is the latest joint video coding standard proposed by ITU-T SG 16 WP and ISO/IEC JTC 1/SC29/WG 11. In H.265, pictures are divided into a sequence of coding tree units(CTUs), and the CTU further is partitioned into multiple CUs to adapt to various local characteristics. Its coding efficiency is approximately two times high compared to previous standard H.264/AVC. However according to the size of extended CU(coding unit) and transform block, the hardware size of PMR(prediction/mode decision/reconstruction) block within video encoder is about 4 times larger than previous standard. In this study, we propose a new less complex hardware architecture of PMR block which has the most high complexity within encoder without any noticeable PSNR loss. Using this simplified block, we can shrink the overall size the H.265 encoder. For FHD image, it operates at clocking frequency of 300 MHz and frame rate of 60 fps. And also for the test image, the Bjøntegaard Delta (BD) bit rate increase about average 30 % in PMR prediction block, and the total estimated gate count of PMR block is around 1.8 M.

• Journal of Biomedical Engineering Research
• /
• v.23 no.1
• /
• pp.49-60
• /
• 2002

Receive dynamic focusing with an array transducer can provide near optimum resolution only in the vicinity of transmit focal depth. A customary method to increase the depth of field is to combine several beams with different focal depths, with an accompanying decrease in the frame rate. In this Paper. we Present a simultaneous multiple transmit focusing method in which chirp signals focused at different depths are transmitted at the same time. These chirp signals are mutually orthogonal in a sense that the autocorrelation function of each signal has a narrow mainlobe width and low sidelobe levels. and the crossorelation function of any Pair of the signals has values smaller than the sidelobe levels of each autocorrelation function. This means that each chirp signal can be separated from the combined received signals and compressed into a short pulse. which is then individually focused on a separate receive beamformer. Next. the individually focused beams are combined to form a frame of image. Theoretically, any two chirp signals defined over two nonoverlapped frequency bands are mutually orthogonal In the present work. however, a tractional overlap of adjacent frequency bands is permitted to design more chirp signals within a given transducer bandwidth. The elevation of the rosscorrelation values due to the frequency overlap could be reduced by alternating the direction of frequency sweep of the adjacent chirp signals We also observe that the Proposed method provides better images when the low frequency chirp is focused at a near Point and the high frequency chirp at a far point along the depth. better lateral resolution is obtained at the far field with reasonable SNR due to the SNR gain in Pulse compression Imaging .