• 제어로봇시스템학회:학술대회논문집
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• 2001.10a
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• pp.98.2-98
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• 2001

It is necessary to have enough transmission capacity for advanced internet techniques and various digital home appliances networking. Home appliances interface IEEE1394 technique has much wealthy transmission skill. IEEE1394 is using home appliances through various information form image and voice change data real time print out. In this paper, AVC CTS technology and IEEE1394 technology are introduced. Digital Video Camera includes compression format using DV. System composition control is consisted of protocols like IEC-61883 and AV/C command set standard.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1986.04a
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• pp.102-106
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• 1986

A digital telephone set for a voice/data PABX has been implemented with 3 semicustom IC`s, and one-chip microprocessor, etc. It adopts TCM(Time Compression Multiplexing) digital transmission method, and has the capacity of 144kb/s full-duplex digital transmission.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.15 no.4
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• pp.161-167
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• 2015

Today, computer network infrastructure is commonplace and internet user increase. Transmission bandwidth for network traffic is increasing. The data amount of 2014 and 2015 year measured 47 and 62 exabyte, it shows a tremendous increase. Video traffic occupy 60 percent among internet traffic. Enormous growth of Internet traffic than the network infrastructure is not enough to support them, so mobile network occur side-effects such as fall of voice communication. In this paper, we proposed the video compression method using SURF feature and block shaping in order to avoid potential damage. Our proposed method can provide enough performance than previous method.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.8
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• pp.1696-1702
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• 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 KIEE Conference
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• 1996.07b
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• pp.1295-1297
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• 1996

CCITT G.721, G.723 standard ADPCM algorithm is implemented by using TI's fixed point DSP start kit (DSK). ADPCM can be implemented on a various rates, such as 16K, 24K, 32K and 40K. The ADPCM is sample based compression technique and its complexity is not so high as the other speech compression techniques such as CELP, VSELP and GSM, etc. ADPCM is widely applicable to most of the low cost speech compression application and they are tapeless answering machine, simultaneous voice and fax modem, digital phone, etc. TMS320C50 DSP is a low cost fixed point DSP chip and C50 DSK system has an AIC (analog interface chip) which operates as a single chip A/D and D/A converter with 14 bit resolution, C50 DSP chip with on-chip memory of 10K and RS232C interface module. ADPCM C code is compiled by TI C50 C-compiler and implemented on the DSK on-chip memory. Speech signal input is converted into 14 bit linear PCM data and encoded into ADPCM data and the data is sent to PC through RS232C. The ADPCM data on PC is received by the DSK through RS232C and then decoded to generate the 14 bit linear PCM data and converted into the speech signal. The DSK system has audio in/out jack and we can input and out the speech signal.

• Journal of Korea Entertainment Industry Association
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• v.13 no.2
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• pp.93-100
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• 2019

Dynamic Compressors in vocal recordings of modern pop music are essential equipment. Dynamic compressors are applied not only to the mix for listening to music but also to the monitor for the singer to listen to his voice along with the accompaniment while the singer is recording. This study is an experimental study on the effects of a dynamic compressor applied to a monitor environment on the vocal performance of a singer. 10 participating singers participated in the blind test to test how the vocals heard through the monitor would be affected by the 1:1, 2:1 and 4:1 compression ratio. Experimental results show that the higher the compression ratio applied to the monitor, the bigger the song, the brighter the tone, but the pitch becomes finer inaccuracy on the bigger dynamic part of the song. In post-interviews with blinds, it was found that singers generally preferred to hear compressed sound through a compressor on the monitor. Since the music used in the experiment was a ballad with a wide dynamic range, it could not be generalized to all kind of music recordings, but it could provide important implications for the monitoring of recording sites. In addition, We hope that the cognitive science approach to recording technology will be added based on this paper which has been studied through empirical studies on the effect of the monitor environment on the singing voice.

• Journal of IKEEE
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• v.5 no.1 s.8
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• pp.8-15
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• 2001

This paper remarks IETF based requirements for IP/UDP/RTP header compression issued in 3GPP2 All IP Ad Hoc Meeting and protocol stacks of the next generation mobile station. All IP Network, for real time application such as Voice over IP (VoIP) multimedia services based on 3GPP2 3G cdma2000. Frames for various protocols expected in the All IP network Mobile Station (MS) are explained with several figures including the bit-for-bit notation of header format based on IETF draft of Robust Header Compression Working Group (ROHC). Especially, this paper includes problems of IS-707 Radio Link Protocol (RLP) for header compression which will be expected to modify in All IP network MS's medium access layer to accommodate real time packet data service[1]. And also, since PPP has also many problems in header compression and mobility aspects in MS protocol stacks for 3G cdma2000 packet data network based on Mobile IP (PN-4286)[2], we introduce the problem of solution for header compression of PPP. Finally. we suggest the guidelines for All IP network MS header compression about expected protocol stacks, radio resource efficiency and performance.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2014.06a
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• pp.157-159
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• 2014

In this paper, we introduce SlimSRP, an ultra-low-power digital signal processor (DSP) solution for mobile audio and voice applications. So far, application processors (APs) have taken charge of all the tasks in mobile devices. However, they have suffered from short battery life problems to deal with complex usage scenarios, such as always-on voice trigger with continuous audio playback. From extensive analysis of audio and voice application characteristics, SlimSRP is designed to relive the performance and power burden of APs. It employs three-issue VLIW architecture, and the major low-power and high-performance techniques include: (1) an optimized register-file architecture friendly for constants generation, (2) a powerful instruction set to reduce the number of register file accesses and (3) a unique instruction compression scheme that contributes to saved memory size and reduced cache miss. An implementation of SlimSRP runs at up to 200MHz and the logic occupies 95K NAND2 gates in Samsung 28LPP process. The experimental results demonstrate that a MP3 decoder application with a 128kbps 44.1kHz input can run at 5.1MHz and the logic consumes only 22uW/MHz.

• Journal of Korea Multimedia Society
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• v.12 no.2
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• pp.257-271
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• 2009

In this paper, we propose a protocol compression scheme for enhancing the performance of call processing of PTT(Push-to-Talk) which is one of the important services in IMS(IP Multimedia Subsystem), a next generation integrated wired/wireless packet communication network. To service the PTT on an IMS network, it should use the same call setup procedure as legacy Mobile and TRS(Trunked Radio System) networks and have a fast call setup time and enough communication bandwidth because a number of terminals should be able to exchange same data in real time. The proposed A+SigComp scheme reduces the initial call setup delay of SIP by about 10%, which is used by PTT service for call setup. In addition, the A+ROHC scheme is proposed to compress the header of RTP packets transferred during PTT voice transmission and, as a result, about 5% of increase in communication efficiency is observed.

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