• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.9 no.3
• /
• pp.493-497
• /
• 2005

This paper is about the research of the User Agent implementation under wireless embedded environment, using SIP which is one of protocol components construct the VoIP system. The User Agent is made of the User Agent configuration block, the device thread block to control devices and the SIP stack block to process SIP messages. The device thread consists of the RTP thread and the sound lard device processing block. Futhermore, the SIP stack consist of the worker thread to process proxy events, the SIP transceiver and SIP thread to transfer and receive SIP messages. The H/W platform is a board included the Intel's XScale PXA255 processor, flash memory, SDRAM, Audio CODEC module and wireless LAN threough PCMCIA socket, furthermore a microphone and headphone is used by the audio 1/0. The system has embedded linux kernel 2.4.19. For embedded environment, the function of User Agent and SIP method is diminished. Finally, the resource of system could be reduced about $12.9\%$, compared to overall system resource, by minimizing peripherals control and excepting TCP.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.37 no.1C
• /
• pp.63-75
• /
• 2012

This paper researched the operation methods of software video teleconferencing on the tactical communication networks under 1Mbps. The tactical communication networks have limited bandwidths, frequent data losses and transmission delays due to the unstable networks. In addition, the bandwidth for video teleconferencing has to be much smaller since the Army Tactical Command Information System(ATCIS) has priority of using the bandwidth. This paper analyzed such restrictions of tactical communication networks, presented some methods to improve the quality of the software video teleconferencing on the tactical communication networks and their actual experiments as well. It is applied in the first place to re-transmit the lost packets and to reduce the image size for the data traffic. Nothing is better for the video teleconferencing than to provide the bandwidth enough for every user. However, on the tactical communication networks with the limited bandwidth, video teleconferencing can be improved by optimizing the compression rate of image data, the number of image frames, the audio codec and the usage of audio compensation data.

• Journal of IKEEE
• /
• v.24 no.4
• /
• pp.1122-1128
• /
• 2020

Bandwidth Extension refers to restoring and expanding a narrow band signal(NB) that is damaged or damaged in the encoding and decoding process due to the lack of channel capacity or the characteristics of the codec installed in the mobile communication device. It means converting to a wideband signal(WB). Bandwidth extension research mainly focuses on voice signals and converts high bands into frequency domains, such as SBR (Spectral Band Replication) and IGF (Intelligent Gap Filling), and restores disappeared or damaged high bands based on complex feature extraction processes. In this paper, we propose a model that outputs an bandwidth extended signal based on an autoencoder among deep learning models, using the residual connection of one-dimensional convolutional neural networks (CNN), the bandwidth is extended by inputting a time domain signal of a certain length without complicated pre-processing. In addition, it was confirmed that the damaged high band can be restored even by training on a dataset containing various types of sound sources including music that is not limited to the speech.

• The Journal of the Acoustical Society of Korea
• /
• v.28 no.7
• /
• pp.589-598
• /
• 2009

As mobile devices become multi-functional, and converge into a single platform, there is a strong need for a codec that is able to provide consistent quality for speech and music content MPEG-D USAC standardization activities started at the 82nd MPEG meeting with a CfP and approved WD3 at the 88th MPEG meeting. MPEG-D USAC is converged technology of AMR-WB+ and HE-AAC V2. Specifically, USAC utilizes three core codecs (AAC ACELP and TCX) for low frequency regions, SBR for high frequency regions and the MPEG Surround tool for stereo information. USAC can provide consistent sound quality for both speech and music content and can be applied to various applications such as multi-media download to mobile device Digital radio Mobile TV and audio books.

• ETRI Journal
• /
• v.25 no.5
• /
• pp.328-336
• /
• 2003

Bluetooth is a specification for short-range wireless communication using the 2.4 GHz ISM band. It emphasizes low complexity, low power, and low cost. This paper describes an area-efficient digital baseband module for wireless technology. For area-efficiency, we carefully consider hardware and software partitioning. We implement complex control tasks of the Bluetooth baseband layer protocols in software running on an embedded microcontroller. Hardware-efficient functions, such as low-level bitstream link control; host controller interfaces (HCIs), such as universal asynchronous receiver transmitter (UART) and universal serial bus (USB)interfaces; and audio Codec are performed by dedicated hardware blocks. Furthermore, we eliminate FIFOs for data buffering between hardware functional units. The design is done using fully synthesizable Verilog HDL to enhance the portability between process technologies so that our module can be easily integrated as an intellectual property core no system-on-a-chip (SoC) ASICs. A field programmable gate array (FPGA) prototype of this module was tested for functional verification and realtime operation of file and bitstream transfers between PCs. The module was fabricated in a $0.25-{\mu}m$ CMOS technology, the core size of which was only 2.79 $mm{\times}2.80mm$.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.17 no.7
• /
• pp.1633-1638
• /
• 2013

The mobile phone has now evolved into an effective multimedia devices to watch video content with your PC in addition to the calling features. Thus, the effectiveness of the video content streaming services smartphone will be available. And content should be able to deliver effectively. Be provided with textbook images and video of the speaker means that the effective content delivery. In this paper, we propose a integrated video management system that can be real-time VOD services on the Internet as input Multi-Source of audio-video, video content encoding system to meet the requirements of the above two.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2013.05a
• /
• pp.590-592
• /
• 2013

Devices compliant to the Universal Plug and Play (UPnP) Audio Video (A/V) specifications can easily establish multimedia home networks through zero or auto configuration. Despite the existing UPnP certification process many issues concerning interoperability still arise. Capabilities may not be fully implemented or there may be codec issues. Therefore, a means to check and verity interoperability prior to product purchasing or usage is necessary. In this paper, an implementation of a portable control point is proposed to store and manage UPnP device capability information. The portable control point can be used to find UPnP A/V compliant devices that are most compatible.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.9 no.8
• /
• pp.1696-1702
• /
• 2005

Real Time Transport Protocol (RTP) is the Internet standard protocol for transport of real time data audio/video IP Telephony, Multimedia Seivece. In case of 8kbps voice codec, the size of packet per data is 20bytes and become more large to minimal 40bytes with adding each layer's header in RTP/UDP/IP. To solve this problem, various header compression skill were suggested on point-to-point networks. But it compress even IP header and cannot be suitable to apply to end-to-end network Thus, We will renew header compression protocol to apply wired router-based network.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2006.05a
• /
• pp.906-909
• /
• 2006

This paper proposes a new body-effect compensated switch configuration for low voltage and low distortion switched-capacitor (SC) applications. The proposed circuit allows rail-to-rail switching operation for low voltage SC circuits and has better total harmonic distortion than the conventional bootstrapped circuit by 19 dB. A 2-1 cascaded sigma-delta modulator is provided for performing the high-resolution analog-to-digital conversion on audio codec in a communication transceiver. An experimental prototype for a single-stage folded-cascode operational amplifier (opamp) and a 2-1 cascaded sigma-delta modulator has been implemented in a 0.25 micron double-poly, triple-metal standard CMOS process with 2.7 V of supply voltage.