• Journal of Korea Multimedia Society
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• v.12 no.5
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• pp.653-663
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• 2009

Synchronization between media is an important aspect in the design of multimedia communication-system. This paper proposes a precise media synchronization mechanism for video and audio transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, RTP/RTCP protocol suite is usually employed. To provide a precise mechanism for media synchronization between video and audio, we suggest an efficient media synchronization algorithm based on NPT (Normal Play Time) which can be derivable from the timestamp information in the header part of RTP packet generated for the transport of video and audio. In the proposed method, we do not need to send and process any RTCP SR (sender report) packet which is required for conventional media synchronization scheme, and accordingly could reduce the number of required UDP ports and the amount of control traffic injected into the network.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.16 no.2
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• pp.69-76
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• 2023

A high-quality audio/voice codec using the MDCT/IMDCT process can perfectly restore the current frame through an overlap-add process with the previous frame. In the overlap-add process, an algorithm delay equal to the frame length occurs. In this paper, we propose a MDCT/IMDCT process that reduces algorithm delay by using a variable phase shift in MDCT/IMDCT process. In this paper, a low-delay audio/speech codec was proposed by applying the low delay MDCT/IMDCT algorithm to the ITU-T standard codec G.729.1 codec. The algorithm delay in the MDCT/IMDCT process can be reduced from 20 ms to 1.25 ms. The performance of the decoded output signal of the audio/speech codec to which low-delay MDCT/IMDCT is applied is evaluated through the PESQ test, which is an objective quality test method. Despite of the reduction in transmission delay, it was confirmed that there is no difference in sound quality from the conventional method.

• The Journal of the Acoustical Society of Korea
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• v.23 no.3
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• pp.262-267
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• 2004

AMR-WB (Adaptive Multi Rate Wideband) speech codec, the most recent voice codec standardized by 3GPP, has the wider audio bandwidth of 50∼7000 Hz and operates on nine speech coding bit rates between 6.60 and 23.85 kbit/s. This Paper presents the real-time implementation of AMR-WB speech codec by using a 16 bit fixed-point TeakLite DSP. The implemented AMR-WB codec requires the complexity of 52.2 MIPS at 23.85 kbit/s mode and also needs the program memory of 17.9 kwords, data RAM of 11.8 kwords, and data ROM of 10.1kwords. It was verified through passing the all test vectors provided by 3GPP with maintaining bit exactness. Stable operations on the real-time testing board were also proved without any distortions and delays for the audio in/out.

• The KIPS Transactions:PartA
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• v.15A no.6
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• pp.317-324
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• 2008

In this paper, we propose a system which easily controls the existing serially connected multi-channel speakers in a general personal computer by using a USB(Universal Serial Bus) interface. The personal computer as a host of the USB interface analyzes a sound source and sends audio data in a real-time fashion by the use of the isochronous transmission, one of four transmission methods provided by the USB interface. In addition, a channel is assigned by means of the bulk transmission, one of four transmission methods provided by the USB interface. Transmitted data from the USB host are sent to each speaker through compression and packet generation process. Each speaker detects corresponding digital data and regenerates audio signals through DAC(Digital-to-Analog Converter). A user can easily select a sound source file and a channel by the use of a GUI environment in a personal computer.

• Journal of the Institute of Electronics and Information Engineers
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• v.49 no.11
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• pp.149-156
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• 2012

A digital input class-D audio amplifier is presented for digital hearing aid. The class-D audio amplifier is composed of digital and analog circuits. The analog circuit converts a digital input to a analog audio signal (DAC) with noise suppression in the audio band. An interpolated digital delta-sigma modulator is used to convert data types between digital signal processor (DSP) and digital-to-analog converter (DAC). An 16-bit, 25-kbps pulse code modulated (PCM) input is interpolated to 16-bit, 50-kbps by a digital filter. The output signal of interpolation filter is noise-shaped by a third-order digital sigma-delta modulator (SDM). As a result, 1.5-bit, 3.2-Mbps signal is applied to simple digital to analog converter.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.17 no.10
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• pp.1063-1073
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• 1992

This paper deals with a method for extracting picture signals from CD graphics with a conventional CD player, schemes for designing circuits for the effective extraction of control signals, and the implementation of such circuits using commercially available logic components, thereby achieving cost-effectiveness. This paper also presents an implementation and evaluation of the CD-G system, which requires extracting picture signals, deinterleaving the extracted signals and analyzing control commands and displaying them on a screen. The CD-G system implemented using the extraction circuit presented herein has been observed to operate well in real time.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2005.11a
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• pp.278-281
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• 2005

본 논문에서는 백색 잡음이 첨가된 오디오 신호로부터 잡음을 추정하고 이에 따른 지각 필터 보정을 통한 적응적 지각 필터의 속도 개선 알고리즘을 제안한다. 제안된 잡음 추정 알고리즘은 고주파 구간에서 획득한 잡음 에너지를 사용하여 지각필터를 구성하고 일정 대역에서 추정된 기울기를 고주파 영역에 적용함으로써 전 대역의 잡음 에너지를 효과적으로 제거할 수 있게 되며 필터의 속도 향상에도 크게 기여하게 된다. 이는 기존의 적응지각필터와의 비교를 위하여 수행속도 및 SSNR, NMR 비교를 수행하였고 그 결과를 통하여 성능 개선을 확인할 수 있었다.

• Broadcasting and Media Magazine
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• v.19 no.1
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• pp.48-55
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• 2014

One of the recent noticeable evolutions in mobile communication systems is that wideband-codec is deployed rapidly in VoLTE (Voice over Long Term Evolution) service or HD voice. This paper is concerned with next generation HD voice or VoLTE service that is coined to describe high quality communication with super-wideband voice codec. 3GPP EVS (Enhanced Voice Service) Codec is being standardized to develop the super-wideband voice codec. This paper deals with the codec design constraints, performance requirements, the status of standardization, and finally perspective on VoLTE service in future.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2016.06a
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• pp.224-227
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• 2016

본 논문에서는 스테레오 채널 신호 간 강도비를 이용하여 음원을 분리하는 EADRess 알고리즘과 부분기반 표현을 특징으로 한 비음수 행렬 인수분해를 통해 음원을 분리하는 NMF 가 결합된 새로운 음원분리 알고리즘을 제안한다. 입력 오디오 신호로부터 frequency-azimuth 평면 구성을 통해 식별된 방위각에 상응하는 신호 강도비로 표현되는 확률밀도함수를 이용하여 1 단계 음원분리를 수행하고, 얻어진 개별 분리음원을 대상으로 supervised NMF 및 Wiener 필터 기반 마스킹 함수를 적용함으로써 잔류 혼합성분을 제거하는 2 단계 음원분리를 수행한다. 제안된 EADRess/NMF 결합 음원분리 알고리즘의 성능을 검증하기 위하여 SASSEC 에서 제공하는 테스트 음원을 이용하여 측정한 결과, 개별 음원분리 알고리즘에 비해 SIR 이 각각 1.41dB, 10.43dB 향상된 결과를 얻었다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2013.11a
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• pp.151-154
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• 2013

다수의 음원이 특정한 공간에 산재하고 있을 때, 그 중 특정 음원에 주목하면 다른 음원과 분리되어 특정 음원만 들리는 현상을 칵테일파티 현상이라고 한다. 심리적인 이 현상에 영감을 받아 음원을 분리하는 알고리즘이 만들어졌다. 이런 음원 분리방법을 Blind Source Separation(BSS) 이라고 하는데, 여러 신호가 섞이는 과정을 모르는 상태에서 음원을 분리한다는 뜻에서 Blind Source Separation 이라고 한다. BSS에 사용되는 알고리즘으로 주로 PCA, ICA이 있다. PCA는 2차원의 경우를, ICA는 그 이상의 고차원의 통계적 특성을 이용한다. 이에 본 논문은 PCA를 이용하여 두 음원을 분리하는 알고리즘을 구현하는데 역점을 두었다. PCA는 주로 음원보다는 이미지 신호 처리에 초점이 맞추어져 있지만, 음원 분리에 있어서도 충분한 성능을 보여주므로, ICA를 이용한 음원 분리 알고리즘과의 비교를 통하여 장, 단점을 알아보고 추후 PCA의 응용 가능성을 알아보았다.