• Journal of Broadcast Engineering
• /
• v.19 no.6
• /
• pp.819-825
• /
• 2014

In this paper, monaural ambient component extraction algorithm based on nonnegative matrix factorization (NMF) is described. The ambience component extraction algorithm in this paper is developed for audio upmixing system; Recent researches have shown that they can enhance listener envelopment if the extracted ambient signal is applied into the multichannel audio upmixing system. However, the conventional method stores all of the audio signal and processes all at once, so it cannot be applied to streaming system and digital signal processor (DSP) system. In this paper, the ambient component extraction algorithm based on on-line nonnegative matrix factorization is developed and evaluated to solve the problem. As a result of analysis of the processed signal with spectral flatness measures in the experiment, it was shown that the developed system can extract the ambient signal similarly with the conventional batch process system.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.31 no.9B
• /
• pp.838-847
• /
• 2006

In order to support multiple multimedia streaming services in home networks, many critical issues must be considered. In addition, handling the shortage of network bandwidth is one of the most significant and complicated issues. In this paper, real-time bandwidth management service is suggested as a solution to the problem regarding the IEEE1394-based home network. In order to handle the shortage of network bandwidth and to enhance the bus utilization rate, the proposed service combines two methods. First, the bus bandwidth management function determines the state of the network bandwidth and restores the residual bandwidth, which is excessively occupied by a streaming service, to the available free bandwidth. Second, the Isochronous Streaming (IS) Scheduler manages all streaming services according to priority. In order to test the proposed service, we implemented a prototype steaming management middleware and evaluated it by using the IEEE1394 network test-bed.

• Journal of KIISE:Computer Systems and Theory
• /
• v.28 no.5
• /
• pp.213-217
• /
• 2001

Excessive memory buffer requirement in continuous media playback is a serious impediment of wide spread usage of on-line multimedia service. Skewed access frequency of available video files provides an opportunity of re-using the date blocks which has been loaded by one session for later usage. We present novel algorithm which minimizes the buffer requirement in multiple sessions of multimedia playbacks. In continuous media playback originated from the disk, a certain amount of memory buffer is required to synchronize asynchronous disk. Read operation and synchronous playback operation. As aggregate playback bandwodth increases, larger amount of buffer needs to be allocated for this synchronization purpose. The focus of this work is to study the asymptotic behavior of the synchronization buffer requirement and to develop an algorithm coping with this excessive buffer requirement under bandwidth congestioon. We argue that in a large scale continuous media server, it may not be necessary to read the blocks for each session directly from the disk. The beauty of our work lies in the fact that it dynamically adapts to disk utilization of the server and finds the optimal way of servicinh the individual sessions while minimizing the overall buffer space requirement. Optimality of the proposed algorithm is shown by proof. The effectiveness and performance of the proposed scheme is examined via simulation.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.34 no.11B
• /
• pp.1151-1168
• /
• 2009

Improving packet loss does not necessarily coincide with the improvement in user perceivable QoS because each frame carries different degree of importance. We propose Significance-aware packet scheduling (SAPS) to maximize user perceivable QoS. SAPS carries out two fundamental issues of packet scheduling: "What to transmit" and "When to transmit?" To adapt to the available bandwidth, it is necessarily to transmit the subset of the data packets if the entire set of packets can not be transmitted. "Packet Significance" quantifies the importance of the frame by elaborately incorporating frames' dependency. Greedy approach is used in selecting packets and transmission schedule is determined based on the Packet Significance. The proposed scheme is tested using publicly available MPEG-4 video clips. Decoding engine is embedded in the simulation software and user perceivable QoS is exposeed in termstermiSNR. Throughout the simulation based experiment, the performance of the proposed scheme is compared two other schemes: Size-based packet scheduling and Bit-rate based best effort packet scheduling. SAPS successfully incorporates the semantics of a packet and improves user perceivable QoS significantly. It successfully provides unequal protection to more important packets.

• Smart Media Journal
• /
• v.9 no.4
• /
• pp.9-16
• /
• 2020

Recently, various studies are being conducted to integrate Image Segmentation into smart factory industries and autonomous driving fields. In particular, Image Segmentation systems using deep learning algorithms have been researched and developed enough to learn from large volumes of data with higher accuracy. In order to use image segmentation in the autonomous driving sector, sufficient amount of learning is needed with large amounts of data and the streaming environment that processes drivers' data in real time is important for the accuracy of safe operation through highways and child protection zones. Therefore, we proposed a novel DFCN algorithm that enhanced existing FCN algorithms that could be applied to various road environments, demonstrated that the performance of the DFCN algorithm improved 1.3% in terms of "loss" value compared to the previous FCN algorithms. Moreover, the proposed DFCN algorithm was applied to the existing U-Net algorithm to maintain the information of frequencies in the image to produce better results, resulting in a better performance than the classical FCN algorithm in the autonomous environment.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.31 no.5B
• /
• pp.430-441
• /
• 2006

As the various A/V devices and home networks are delivered to users, home networks are changing to an entertainment network. It is expected that the required network bandwidth and the amount of usage of media content in home entertainment networks will be increased. Although the access networks and home networks becoming a high speed network, there remains the problems for QoS-guaranteed media content transfer in home networks. Also, in the home network, there can be network traffic caused by applications like video conferencing, video telephone, and VoIP(voice over IP) as well as inner network traffic of home network. Since media content transfer requires the real-time delivery, it is very important and basic requirement that is to transfer media content to A/V device user wants while keeping the media quality. Even though there are many middleware protocol for home networking, they provide basic device discovery and control or simple functions for QoS-guaranteed media content transfer that are not enough to provide QoS-guaranteed media transfer service that user wants. Thus, in this paper, we propose the technique based on UPnP(universal plug and play) protocol for QoS-guaranteed media content transfer in the home network. The proposed technique is compatible with UPnP and can be used with UPnP as additional functions. In this paper, we utilize VideoLAN application to verify the proposed technique. We add the additional modules that support the proposed technique's function to VideoLAN and we verify the its functions through various test scenarios.

• Journal of Broadcast Engineering
• /
• v.25 no.3
• /
• pp.305-312
• /
• 2020

As a demand for a new video coding standard having higher coding efficiency than the existing standards is growing, recently, MPEG and VCEG has been developing and standardizing the next-generation video coding project, named Versatile Video Coding (VVC). Many inter prediction techniques have been introduced to increase the coding efficiency, and among them, an adaptive motion vector resolution (AMVR) technique has contributed on increasing the efficiency of VVC. However, the best motion vector can only be determined by computing many rate-distortion costs, thereby increasing encoding complexity. It is necessary to reduce the complexity for real-time video broadcasting and streaming services, but it is yet an open research topic to reduce the complexity of AMVR. Therefore, in this paper, an efficient technique is proposed, which reduces the encoding complexity of AMVR. For that, the proposed method exploits a special VVC tree structure (i.e., multi-type tree structure) to accelerate the decision process of AMVR. Experiment results show that the proposed decision method reduces the encoding complexity of VVC test model by 10% with a negligible loss of coding efficiency.

• Journal of Broadcast Engineering
• /
• v.25 no.4
• /
• pp.609-619
• /
• 2020

In accordance with a demand for high quality video streaming, it needs high data rate in limited bandwidth and more traffic congestion occurs. In particular, when providing real time video service, packet loss rate and bit error probability increase significantly. To solve these problems, a raptor code, which is one of FEC(Forward Error Correction) techniques, is pervasively used in the application layers as a method for improving real-time service quality. In this paper, we propose a method of determining image transmission parameters based on various deep neural networks to increase transmission efficiency at a similar level of image quality by using raptor codes. The proposed neural network uses the packet loss rate, video encoding rate and data rate as inputs, and outputs raptor FEC parameters and packet sizes. The results of the proposed method present that the throughput is 1.2% higher than that of the existing multimedia transmission technique by optimizing the transmission efficiency at a PSNR(Peak Signal-to-Noise Ratio) level similar to that of the existing technique.

• Journal of Popular Narrative
• /
• v.27 no.2
• /
• pp.107-142
• /
• 2021

Emergence of new media platform had changed relationship between the broadcaster and the viewer, which used to form 'performer-audience' structure. This research has focused on the transition of 'streamer-viewer' role in single-media broadcasting, such as Youtube or Twitch, and identify how they progress group storytelling as a team. Walter Benjam and Leslie Marmon Silko's notion of 'story and storyteller' and Erving Goffman's 'social role theory' was used to define participants' role in new media broadcasting. channel, on Youtube, was selected and analyzed as example case. The domain of 'front stage' was broadened in recorded contents comparing to live streaming. The audience of live streaming is included to the front stage during the expansion. The role of streamer, game participant, and live stream contents viewer is also adjusted during the change, which leads to group-creation of the contents. Streamer plays a role of main-storyteller and suggest identity of the community. Game participants work as sub-storyteller, filling in the blank space in game storytelling and making it sophisticated. They also perform based on community's identity, which streamer has built in advance. Lastly, live steam viewers are intermittent sub-storyteller, which seldom add up the narrative. Though, their main role is to preserve identity of game broadcasting community by reacting according to community's identity. As a result, the game broadcasting narrative is developed by combining and adding up pieces of story made in different level and role of participants. The research redefine the role of viewer and storytelling method in new media, especially in single-person broadcasting. Considering the rapid shift in recent media and contents, a new approach to the streamer-veiwer role and group storytelling of this research can be one of the new method to analyze contents produced in new media, such as Youtube.

• Journal of Korea Multimedia Society
• /
• v.14 no.1
• /
• pp.54-75
• /
• 2011

PoC(Push-to-talk Over Cellular) is an integrated technology of group voice calls, video calls and internet based multimedia services. If a PoC user can not participate in the PoC session for various reasons such as an emergency situation, lack of battery capacity, then the user can use the PoC Box which has a similar functionality to the MM Box in the MMS(Multimedia Messaging Service). The RTSP(Real-Time Streaming Protocol) method is recommended to be used when there is a transmission session between the PoC box and a terminal. Since the existing VOD service uses a wired network, the packet size of RTSP-based VOD service is huge, however, the PoC service has wireless communication environments which have general characteristics to be used in RTSP method. Packet loss in a wired communication environments is relatively less than that in wireless communication environment, therefore, a buffering latency occurs in PoC service due to a play-out delay which means an asynchronous play of audio & video contents. Those problems make a user to be difficult to find the information they want when the media contents are played-out. In this paper, the following techniques and methods were proposed and their performance and superiority were verified through testing: cross-over dual reception buffering technique, advance partition multi-reception buffering technique, and on-demand multi-reception buffering technique, which are designed for effective picking up of information in media content being transmitted in short amount of time using RTSP when a user searches for media, as well as for reduction in playback delay; and same-priority packetization transmission method and priority-based packetization transmission method, which are media data packetization methods for transmission. From the simulation of functional evaluation, we could find that the proposed multiple receiving buffering and packetizing methods are superior, with respect to the media retrieval inclination, to the existing single receiving buffering method by 6-9 points from the viewpoint of effectiveness and excellence. Among them, especially, on-demand multiple receiving buffering technology with same-priority packetization transmission method is able to manage the media search inclination promptly to the requests of users by showing superiority of 3-24 points above compared to other combination methods. In addition, users could find the information they want much quickly since large amount of informations are received in a focused media retrieval period within a short time.