• Title/Summary/Keyword: Low-delay audio

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A Systematical Design of ADSL POTS Splitter Using Passive Devices (수동 소자를 이용한 ADSL POTS Splitter의 체계적인 설계)

  • 박지만;김진태;소운섭
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.24 no.6A
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    • pp.913-919
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    • 1999
  • A systematic synthesis process is presented for the design of ADSL POTS splitters. It consists of a low-pass filter formed by a single-ended inductor, balanced inductor, and balance tightly coupled transformer. This three low-pass filters has been simulation. Simulation results show agreement of frequency characteristics. Therefore, POTS splitters using a commercial balance tightly coupled transformer are designed for the applications of ADSL system. The experimental results show that POTS splitter in the ADSL system has ripple decibel of less than $\pm$0.5 dB over a frequency range from 0.2 kHz to 3.4 kHz(or an audio band frequency) and delay distortion of less than 130 $mutextrm{s}$ over a frequency range from 0.6 kHz to 3.2 kHz.

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An Adaptive Multimedia Synchronization Scheme for Media Stream Delivery in Multimedia Communication (멀티미디어 통신에서 미디어스트림 전송을 위한 적응형 멀티미디어 동기화 기법)

  • Lee, Gi-Sung
    • The KIPS Transactions:PartC
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    • v.9C no.6
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    • pp.953-960
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    • 2002
  • Rel-time application programs have constraints which need to be met between media-data. It is client-leading synchronization that is absorbing variable transmission delay time and that is synchronizing by feedback control and palyout control. It is the important factor for playback rate and QoS if the buffer level is normal or not. This paper, The method of maintenance buffer normal state transmits in multimedia server by appling feedback of filtering function. And synchronization method is processing adaptive playout time for smooth presentation without cut-off while media frame is skip. When audio frame which is master media is in upper threshold buffer level we decrease play out time gradually, low threshold buffer level increase it slowly.

Connection Management Scheme using Mobile Agent System

  • Lim, Hee-Kyoung;Bae, Sang-Hyun;Lee, Kwang-Ok
    • Journal of Integrative Natural Science
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    • v.11 no.4
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    • pp.192-196
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    • 2018
  • The mobile agent paradigm can be exploited in a variety of ways, ranging from low-level system administration tasks to middle ware to user-level applications. Mobile agents can be useful in building middle-ware services such as active mail systems, distributed collaboration systems, etc. An active mail message is a program that interacts with its recipient using a multimedia interface, and adapts the interaction session based on the recipient's responses. The mobile agent paradigm is well suitable to this type of application, since it can carry a sender-defined session protocol along with the multimedia message. Mobile agent communication is possible via method invocation on virtual references. Agents can make synchronous, one-way, or future-reply type invocations. Multicasting is possible, since agents can be aggregated hierarchically into groups. A simple check-pointing facility has also been implemented. Another proposed solution is to use multi agent computer systems to access, filter, evaluate, and integrate this information. We will present the overall architectural framework, our agent design commitments, and agent architecture to enable the above characteristics. Besides, the each information needed a mobile agent system such as text, graphic, image, audio and video etc, constructed a great capacity multimedia database system. However, they have problems in establishing connections over multiple subnetworks, such as no end-to-end connections, transmission delay due to ATM address resolution, no QoS protocols. We propose a new connection management scheme in the thesis to improve the connection management involved of mobile agent systems.

A digital Audio Watermarking Algorithm using 2D Barcode (2차원 바코드를 이용한 오디오 워터마킹 알고리즘)

  • Bae, Kyoung-Yul
    • Journal of Intelligence and Information Systems
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    • v.17 no.2
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    • pp.97-107
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    • 2011
  • Nowadays there are a lot of issues about copyright infringement in the Internet world because the digital content on the network can be copied and delivered easily. Indeed the copied version has same quality with the original one. So, copyright owners and content provider want a powerful solution to protect their content. The popular one of the solutions was DRM (digital rights management) that is based on encryption technology and rights control. However, DRM-free service was launched after Steve Jobs who is CEO of Apple proposed a new music service paradigm without DRM, and the DRM is disappeared at the online music market. Even though the online music service decided to not equip the DRM solution, copyright owners and content providers are still searching a solution to protect their content. A solution to replace the DRM technology is digital audio watermarking technology which can embed copyright information into the music. In this paper, the author proposed a new audio watermarking algorithm with two approaches. First, the watermark information is generated by two dimensional barcode which has error correction code. So, the information can be recovered by itself if the errors fall into the range of the error tolerance. The other one is to use chirp sequence of CDMA (code division multiple access). These make the algorithm robust to the several malicious attacks. There are many 2D barcodes. Especially, QR code which is one of the matrix barcodes can express the information and the expression is freer than that of the other matrix barcodes. QR code has the square patterns with double at the three corners and these indicate the boundary of the symbol. This feature of the QR code is proper to express the watermark information. That is, because the QR code is 2D barcodes, nonlinear code and matrix code, it can be modulated to the spread spectrum and can be used for the watermarking algorithm. The proposed algorithm assigns the different spread spectrum sequences to the individual users respectively. In the case that the assigned code sequences are orthogonal, we can identify the watermark information of the individual user from an audio content. The algorithm used the Walsh code as an orthogonal code. The watermark information is rearranged to the 1D sequence from 2D barcode and modulated by the Walsh code. The modulated watermark information is embedded into the DCT (discrete cosine transform) domain of the original audio content. For the performance evaluation, I used 3 audio samples, "Amazing Grace", "Oh! Carol" and "Take me home country roads", The attacks for the robustness test were MP3 compression, echo attack, and sub woofer boost. The MP3 compression was performed by a tool of Cool Edit Pro 2.0. The specification of MP3 was CBR(Constant Bit Rate) 128kbps, 44,100Hz, and stereo. The echo attack had the echo with initial volume 70%, decay 75%, and delay 100msec. The sub woofer boost attack was a modification attack of low frequency part in the Fourier coefficients. The test results showed the proposed algorithm is robust to the attacks. In the MP3 attack, the strength of the watermark information is not affected, and then the watermark can be detected from all of the sample audios. In the sub woofer boost attack, the watermark was detected when the strength is 0.3. Also, in the case of echo attack, the watermark can be identified if the strength is greater and equal than 0.5.

Implementation of QoS-awared MAC Protocol for Home Networks (홈 네트워크를 위한 QOS 보장형 매체접속제어 프로토콜의 구현)

  • 황원주
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.7 no.2
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    • pp.228-238
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    • 2003
  • We believe that existing wire solutions such as HomePNA2.0 using phone lines and HomePlug using power line and wireless solution such as HomeRF are the most promising solutions, because of its cost-effectiveness. However, MAC protocols of these solutions provide only Class of Service(CoS) using priority mechanism like HomePNA and HomePlug or consider only voice among real-time traffics like HomeRF. For these reasons, we perceive the needs of the new MAC protocol which is no new wire solution and provides guaranteed Quality of Service (QoS) for not only voice but also video and audio. In light of this, we present the design and software implementation of a new MAC protocol for Home Networks. Our evaluation results of software implementation verify that proposed MAC protocol can provide low delay, low loss, and low jitter to real-time traffic by reserving bandwidth.

Design of 8K Broadcasting System based on MMT over Heterogeneous Networks

  • Sohn, Yejin;Cho, Minju;Paik, Jongho
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.11 no.8
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    • pp.4077-4091
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    • 2017
  • This paper presents the design of a broadcasting scenario and system for an 8K-resolution content. Due to an 8K content is four times larger than the 4K content in terms of size, many technologies such as content acquisition, video coding, and transmission are required to deal with it. Therefore, high-quality video and audio for 8K (ultra-high definition television) service is not possible to be transmitted only using the current terrestrial broadcasting system. The proposed broadcasting system divides the 8K content into four 4K contents by area, and each area is hierarchically encoded by Scalable High-efficiency Video Coding (SHVC) into three layers: L0, L1, and L2. Every part of the 8K video content divided into areas and hierarchy is independently treated. These parts are transmitted over heterogeneous networks such as digital broadcasting and broadband networks after going through several processes of generating signal messages, encapsulation, and packetization based on MPEG media transport. We propose three methods of generating streams at the sending entity to merge the divided streams into the original content at the receiving entity. First, we design the composition information, which defines the presentation structure for displays. Second, a descriptor for content synchronization is included in the signal message. Finally, we define the rules for generating "packet_id" among the packet header fields and design the transmission scheduler to acquire the divided streams quickly. We implement the 8K broadcasting system by adapting the proposed methods and show that the 8K-resolution contents are stably received and serviced with a low delay.