• Title/Summary/Keyword: 채널 품질

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Performance of Uncompressed Audio Distribution System over Ethernet with a L1/L2 Hybrid Switching Scheme (L1/L2 혼합형 중계 방법을 적용한 이더넷 기반 비압축 오디오 분배 시스템의 성능 분석)

  • Nam, Wie-Jung;Yoon, Chong-Ho;Park, Pu-Sik;Jo, Nam-Hong
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.46 no.12
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    • pp.108-116
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    • 2009
  • In this paper, we propose a Ethernet based audio distribution system with a new L1/L2 hybrid switching scheme, and evaluate its performance. The proposed scheme not only offers guaranteed low latency and jitter characteristics that are essentially required for the distribution of high-quality uncompressed audio traffic, and but also provide an efficient transmission of data traffic on the Ethernet environment. The audio distribution system with a proposed scheme consists of a master node and a number of relay nodes, and all nodes are mutually connected as a daisy-chain topology through up and downlinks. The master node generates an audio frame for each cycle of 125us, and the audio frame has 24 time slotted audio channels for carrying stereo 24 channels of 16-bit PCM sampled audio. On receiving the audio frame from its upstream node via the downlink, each intermediate node inserts its audio traffic to the reserved time slot for itself, then relays again to next node through its physical layer(L1) transmission - repeating. After reaching the end node, the audio frame is loopbacked through the uplink. On repeating through the uplink, each node makes a copy of audio slot that node has to receive, then play the audio. When the audio transmission is completed, each node works as a normal L2 switch, thus data frames are switched during the remaining period. For supporting this L1/L2 hybrid switching capability, we insert a glue logic for parsing and multiplexing audio and data frames at MII(Media Independent Interlace) between the physical and data link layers. The proposed scheme can provide a good delay performance and transmission efficiency than legacy Ethernet based audio distribution systems. For verifying the feasibility of the proposed L1/L2 hybrid switching scheme, we use OMNeT++ as a simulation tool with various parameters. From the simulation results, one can find that the proposed scheme can provides outstanding characteristics in terms of both jitter characteristic for audio traffic and transmission efficiency of data traffics.

Evaluation of a signal segregation by FDBM (FDBM의 음원분리 성능평가)

  • Lee, Chai-Bong
    • The Journal of the Korea institute of electronic communication sciences
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    • v.8 no.12
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    • pp.1793-1802
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    • 2013
  • Various approaches for sound source segregation have been proposed. Among these approaches, frequency domain binaural model(FDBM) has the advantages of low computational load and effective howling cancellation. A binaural hearing assistance system based on FDBM has been proposed. This system can enhance desired signal based on the directivity information. Although FDBM has been evaluated in terms of signal-to-noise ratio (SNR) and coherence function, the evaluation results do not always agree with the human impressions. These evaluation methods provide physical measures, and do not take account of perceptual aspect of human being. Considering a binaural hearing assistance system as a one of major applications, the quality of segregated sound should keep level enough. In the paper, signal segregation performance by means of FDBM is evaluated by three objective methods, i.e., SNR, coherence and Perceptual Evaluation of Speech Quality(PESQ), to discuss the characteristic of FDBM on the sound source segregation performance. The simulation's evaluation results show that FDBM improves the quality of the left and right channel signals to an equivalent level. And the results suggest the possibility that PESQ provides a more useful measure than SNR and coherence in terms of the segregation performance of FDBM. The evaluation results by PESQ show the effects from segregation parameters and indicate appropriate parameters under the conditions. In the paper, signal segregation performance by means of FDBM is evaluated by three objective methods, i.e., SNR, coherence and PESQ, to discuss the characteristic of FDBM on the sound source segregation performance. The simulation's evaluation results show that FDBM improves the quality of the left and right channel signals to an equivalent level. And the results suggest the possibility that PESQ provides a more useful measure than SNR and coherence in terms of the segregation performance of FDBM. The evaluation results by PESQ show the effects from segregation parameters and indicate appropriate parameters under the conditions.

Improved ErtPS Scheduling Algorithm for AMR Speech Codec with CNG Mode in IEEE 802.16e Systems (IEEE 802.16e 시스템에서의 CNG 모드 AMR 음성 코덱을 위한 개선된 ErtPS 스케줄링 알고리즘)

  • Woo, Hyun-Je;Kim, Joo-Young;Lee, Mee-Jeong
    • The KIPS Transactions:PartC
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    • v.16C no.5
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    • pp.661-668
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    • 2009
  • The Extended real-time Polling Service (ErtPS) is proposed tosupport QoS of VoIP service with silence suppression which generates variable size data packets in IEEE 802.16e systems. If the silence is suppressed, VoIP should support Comfort Noise Generation (CNG) which generates comfort noise for receiver's auditory sense to notify the status of connection to the user. CNG mode in silent-period generates a data with lower bit rate at long packet transmission intervals in comparison with talk-spurt. Therefore, if the ErtPS, which is designed to support service flows that generate data packets on a periodic basis, is applied to silent-period, resources of the uplink are used inefficiently. In this paper, we proposed the Improved ErtPS algorithm for efficient resource utilization of the silent-period in VoIP traffic supporting CNG. In the proposed algorithm, the base station allocates bandwidth depending on the status of voice at the appropriate interval by havingthe user inform the changes of voice status. The Improved ErtPS utilizes the Cannel Quality Information Channel (CQICH) which is an uplink subchannel for delivering quality information of channel to the base station on a periodic basis in 802.16e systems. We evaluated the performance of proposed algorithm using OPNET simulator. We validated that proposed algorithm improves the bandwidth utilization of the uplink and packet transmission latency

Study on TV Home Shopping in the Age of broadcast-Communication Fusion (방통융합시대에 TV 홈쇼핑 정책에 관한 연구)

  • Kim, Man-Hwan
    • Journal of Distribution Research
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    • v.16 no.5
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    • pp.19-41
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    • 2011
  • Home shopping is called the flower of future distribution. Among them, for TV home shopping, Korea should lead the industry as an advanced country in IT technology. It is difficult for small and medium businesses, however. This is due to the failure of the policies of Korea Communications Commission. In order to complement such policy failure, this study intends to analyze the preceding domestic studies and foreign status, and reflect such on the policy. Especially the domestic business will be predicted through foreign home shopping status and policy cases, and the items that should be supplemented with governmental policy in order to vitalize small & medium businesses were analyzed; these should be actively reflected by the government. In case of general PP, it already runs on the registration system, and the home shopping channel should also be the same and be committed to the market. The increased number of channels would allow a greater opportunity for small and medium businesses who are the suppliers, and for the consumers, the product prices would be able to be stabilized at an appropriate cost and high quality and diversity of items could be obtained through home shopping, thus improving consumer welfare. If it is difficult to adopt the registration system for home shopping right away, the best alternative is to strengthen the approval system. As home shopping requires a re-approval every 3 years, strengthened re-approval conditions would resolve part of the problem.

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Performance Analysis of MVDR and RLS Beamforming Using Systolic Array Structure (시스토릭 어레이 구조를 갖는 최소분산 비왜곡응답 및 최소자승 회귀 빔형성기법 성능 분석)

  • 이호중;서상우;이원철
    • The Journal of the Acoustical Society of Korea
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    • v.22 no.1
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    • pp.1-6
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    • 2003
  • This paper analyses the performance of either the minimum variance distortionless response (MVDR) or the recursive least square (RLS) beamformer structured on the systolic array. Provided that the snapshot vector including the desired user's signal and the interferences with the noise is received at the array antenna. In order to improve the quality of received signal, MVDR or RLS algorithm can be utilized to update the beamformer weights recursively. Furthermore to increase the channel capacity, by the usage of the above schemes, the effect of the spatial filtering can be obtained which constructively combining multipath components corresponding to the desired user whereas the multiple access interferences (MAI) is nulled out on spatial domain. This paper introduces the MVDR and RLS beamformer structured on systolic array conducting the spatial filtering, and its performance under the multipath fading channel in the presence of multiple access interferences will be analyzed. To show the superior spatial filtering performances of the proposed scheme employing the systolic way structured beamformer, the computer simulations are carried out. And the validity of practical deployment of the proposed scheme will be confirmed throughout showing the BER behaviors and the beampatterns.

An Effective Error-Concealment Approach for Video Data Transmission over Internet (인터넷상의 비디오 데이타 전송에 효과적인 오류 은닉 기법)

  • 김진옥
    • Journal of KIISE:Computing Practices and Letters
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    • v.8 no.6
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    • pp.736-745
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    • 2002
  • In network delivery of compressed video, packets may be lost if the channel is unreliable like Internet. Such losses tend to of cur in burst like continuous bit-stream error. In this paper, we propose an effective error-concealment approach to which an error resilient video encoding approach is applied against burst errors and which reduces a complexity of error concealment at the decoder using data hiding. To improve the performance of error concealment, a temporal and spatial error resilient video encoding approach at encoder is developed to be robust against burst errors. For spatial area of error concealment, block shuffling scheme is introduced to isolate erroneous blocks caused by packet losses. For temporal area of error concealment, we embed parity bits in content data for motion vectors between intra frames or continuous inter frames and recovery loss packet with it at decoder after transmission While error concealment is performed on error blocks of video data at decoder, it is computationally costly to interpolate error video block using neighboring information. So, in this paper, a set of feature are extracted at the encoder and embedded imperceptibly into the original media. If some part of the media data is damaged during transmission, the embedded features can be extracted and used for recovery of lost data with bi-direction interpolation. The use of data hiding leads to reduced complexity at the decoder. Experimental results suggest that our approach can achieve a reasonable quality for packet loss up to 30% over a wide range of video materials.

Equalization Digital On-Channel Repeater Part 1 : Laboratory Test Results (등화형 디지털 동일 채널 중계기 Part 1 : 실험실 테스트 결과)

  • Park Sung Ik;Lee Yong-Tae;Eum Homin;Seo Jae Hyun;Kim Heung Mook;Kim Seung Won;Lee Soo-In
    • Journal of Broadcast Engineering
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    • v.10 no.2
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    • pp.210-220
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    • 2005
  • This paper presents and analyzes laboratory test results of Equalization Digital On-Channel Repeater (EDOCR) using ATSC(Advanced Television Systems Committee) terrestrial digital TV broadcasting system. The EDOCR laboratory test, which is done at CRC(Communications Research Centre) Canada, is classified to receiver test, transmitter test, and synchronization test between transmission and reception frequencies. The receiver part includes feedback signal, random noise, single echo, multi-path ensemble, and NTSC/DTV interference test. The transmitter part includes out-of channel emission, quality of transmitting signal, and phase noise test. By the field test results, the receiver part of the EDOCR eliminates average 5.5 dB of feedback or single echo signal in range of 0 to 11 ${\mu}s$ and has average 18.6 dB at TOV(Threshold of Visibility) under random noise environment. Also, the transmitter part of the EDOCR satisfies the specification of US FCC(Federal Communications Commission), and frequency difference between transmitter and receiver is zero.

A Digital Up-Down Conversion for Wibro Repeater using IIR Filters having Almost Linear Phase Response (유사 선형 위상 특성을 갖는 IIR 필터군을 이용한 Wibro용 디지털 상하향 변환 연구)

  • Chang, Hyung-Min;Lee, Won-Cheol
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.34 no.2C
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    • pp.209-216
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    • 2009
  • The repeater for wireless broadband internet (Wibro) system using OFDM demands the short processing delay to eliminate inter-symbol interference resulted from the time delay greater than the guard time. Towards this, the total system delay of repeater is expected to be minimized as possible as it can without distorting signal quality. In general, the FIR-type of filter is commonly deployed as a channelization filter, but due to its large amount of coefficients for producing prerequisite filter response the excessive large time delay occurs. To withstand this problem, the paper proposes the method for designing IIR filter whose response almost identical to that of the original filter. Moreover, in order to linearize the phase response of the designed IIR filter, this paper also introduce the way of designing the all-pass filter to be cascaded works for linearizing phase response of the channelization as well as the de-channelization filter. To achieve the further improvement in linearization of the phase response and reduction of the overall complexity, this paper tries to transform the integrated IIR filter group into the structure in polyphase style. The computer simulation verifies that the integrated IIR filter group designed in this paper reveals the relatively short processing delay without harming the acceptible signal quality.

A Study on Field Seismic Data Processing using Migration Velocity Analysis (MVA) for Depth-domain Velocity Model Building (심도영역 속도모델 구축을 위한 구조보정 속도분석(MVA) 기술의 탄성파 현장자료 적용성 연구)

  • Son, Woohyun;Kim, Byoung-yeop
    • Geophysics and Geophysical Exploration
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    • v.22 no.4
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    • pp.225-238
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    • 2019
  • Migration velocity analysis (MVA) for creating optimum depth-domain velocities in seismic imaging was applied to marine long-offset multi-channel data, and the effectiveness of the MVA approach was demonstrated by the combinations of conventional data processing procedures. The time-domain images generated by conventional time-processing scheme has been considered to be sufficient so far for the seismic stratigraphic interpretation. However, when the purpose of the seismic imaging moves to the hydrocarbon exploration, especially in the geologic modeling of the oil and gas play or lead area, drilling prognosis, in-place hydrocarbon volume estimation, the seismic images should be converted into depth domain or depth processing should be applied in the processing phase. CMP-based velocity analysis, which is mainly based on several approximations in the data domain, inherently contains errors and thus has high uncertainties. On the other hand, the MVA provides efficient and somewhat real-scale (in depth) images even if there are no logging data available. In this study, marine long-offset multi-channel seismic data were optimally processed in time domain to establish the most qualified dataset for the usage of the iterative MVA. Then, the depth-domain velocity profile was updated several times and the final velocity-in-depth was used for generating depth images (CRP gather and stack) and compared with the images obtained from the velocity-in-time. From the results, we were able to confirm the depth-domain results are more reasonable than the time-domain results. The spurious local minima, which can be occurred during the implementation of full waveform inversion, can be reduced when the result of MVA is used as an initial velocity model.

Hardware optimized high quality image signal processor for single-chip CMOS Image Sensor (Single-chip CMOS Image Sensor를 위한 하드웨어 최적화된 고화질 Image Signal Processor 설계)

  • Lee, Won-Jae;Jung, Yun-Ho;Lee, Seong-Joo;Kim, Jae-Seok
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.44 no.5
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    • pp.103-111
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    • 2007
  • In this paper, we propose a VLSI architecture of hardware optimized high quality image signal processor for a Single-chip CMOS Image Sensor(CIS). The Single-chip CIS is usually used for mobile applications, so it has to be implemented as small as possible while maintaining the image quality. Several image processing algorithms are used in ISP to improve captured image quality. Among the several image processing blocks, demosaicing and image filter are the core blocks in ISP. These blocks need line memories, but the number of line memories is limited in a low cost Single-chip CIS. In our design, high quality edge-adaptive and cross channel correlation considered demosaicing algorithm is adopted. To minimize the number of required line memories for image filter, we share the line memories using the characteristics of demosaicing algorithm which consider the cross correlation. Based on the proposed method, we can achieve both high quality and low hardware complexity with a small number of line memories. The proposed method was implemented and verified successfully using verilog HDL and FPGA. It was synthesized to gate-level circuits using 0.25um CMOS standard cell library. The total logic gate count is 37K, and seven and half line memories are used.