• JSTS:Journal of Semiconductor Technology and Science
• /
• v.15 no.6
• /
• pp.608-614
• /
• 2015

A multi-channel audio CODEC with inter-channel interference suppression is proposed, in which channel switching noise-referred sampling error is significantly reduced. It also supports a coarse/fine mode operation for fast frequency tracking with good harmonic performance. The proposed multi-channel audio CODEC was designed in a 65 nm CMOS process. Measured results indicated that SNR and SNDR of ADC were 93 dB and 84dB, respectively, with SNDR improved by 43 dB. Those of DAC were 96 dB and 87 dB, respectively, with SNDR improved by 45 dB when all the channels are running independently.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 1997.06a
• /
• pp.37-42
• /
• 1997

This paper describes two different S/W-based video codec systems. One is a frame-based video codec with fixed structure based on ITU H.263 standard and the other an object-based video codec with flexible architecture based on ISO MPEG-4 standard currently under specification and planned to be finalized in 1998. These codes are an experimental implementations for examining the feasibility of real-time and/or flexible S/W-based video codecs operating in intra and/or internetworking environments.

• Speech Sciences
• /
• v.11 no.4
• /
• pp.67-73
• /
• 2004

Packet loss, jitter, and delay in the Internet are caused mainly by the shortage of network bandwidth. It is due to queuing and routing process in the intermediate nodes of the packet network. In the Internet whose bandwidth is changing very rapidly in time depending on the number of users and data traffic, controlling the peak transmission bit-rate of a VoIP. system depending on the channel condition could be very helpful for making use of the available network bandwidth. Adapting packet size to the channel condition can reduce packet loss to improve the speech quality. It has been shown in [1] that a VoIP system with an AMR speech codec provides better speech quality than VoIP systems with fixed rate speech codecs. With the adaptive codec mode assignment. algorithm proposed in [1], in this paper, we performed the voice transmission experiments using the wireless LAN through the real Internet environment. Experimental results are analyzed and discussed with our findings.

• Journal of the Korean Institute of Telematics and Electronics
• /
• v.23 no.4
• /
• pp.429-438
• /
• 1986

In this paper we have designed and implemented a real-time, full-duplex SBC (sub-band coding) codec at 16kbps using a high speed digital signal processor, NEC 7720. The SBC codec employs a QMF(quadrature mirror filter) filter bank based on the tree structures of two-band analysis-synthesis pairs to partition speech signal into 4 octabe bands. Computer simulation has been done to investigate the effect of fixed-point computation of the NEC 7720. Three different performance measures, the conventional signal-to-noise ratio, the informal listening test, and an LPC(linear predictive coding)distance measure, have been used in this simulation. The necessary parameters have been optimized through the simulation. The developed hardware and software have been tested in real-time operation using a hardware emulator.

• Proceedings of the KIEE Conference
• /
• 1984.07a
• /
• pp.103-106
• /
• 1984

• Journal of Broadcast Engineering
• /
• v.9 no.3
• /
• pp.236-245
• /
• 2004

A multiview sequence CODEC with view scaiability is proposed in this paper. We define a GGOP (Group of GOP) structure as a basic coding unit to efficiently code multiview sequences. 7he proposed CODEC provides flexible GGOP structures based on the number of views and baseline distances among cameras. Multiview sequences encode consists of disparity estimation/compensation, motion estimation/compensation, residual coding and rate control and generates multiview sequence bitstream. The main bitstream is the same as an MPEG-2 mono-sequence bitstream for MPEG-2 compatibility. The auxiliary bitstream contains information concerning the remaining multiview sequences except for the reference sequences. The proposed CODEC with view scalability provides that a number of view flints are selectively determined at the receiver according to the type of display modes. The proposed multiview sequence CODEC is tested with several multiview sequences to determine its flexibility. compatibility with MPEG-2 and view scaiability. In addition, we subjectively confirm that the decoded bitstreams with view scaiability can be Properly displayed by several types of display modes. including 3D monitors.

• 한국HCI학회:학술대회논문집
• /
• 2008.02a
• /
• pp.218-223
• /
• 2008

In this paper, we implement several methods to improve an audio collaboration system over IP networks, and then evaluate the performance of the implemented methods. In general, speech and audio quality degrades depending on the characteristics of IP networks such as jitter and packet loss. In order to reduce this quality degradation, we propose a lower bit rate audio delivery scheme using the MPEG-2 AAC (Advanced Audio Coding) audio codec in a viewpoint that a packet loss rate could be reduced by a smaller packet size. In addition, iLBC (Internet Low-Bitrate Codec) and the G.711 packet loss concealment algorithm defined by IEFT and ITU-T, respectively, are applied to a audio collaboration system. RAT (Robust-Audio Tool)[7] is used as a baseline platform for the implementation of the proposed methods. It is shown from the implementation that the implemented MPEG-2 AAC audio codec with a bitrate of 256 kbit/s performs as similar as the uncompressed audio quality with a bitrate of 512 kbit/s, and that iLBC and the G.711 packet loss concealment algorithm can improve speech quality when a packet loss rate is 2~10%.

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

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.38 no.4
• /
• pp.353-361
• /
• 2001

A new stereoscopic codec. structure using MPEG-2 multiview profile is presented in this paper. In the suggested codec., the left image is coded with motion estimation in the base layer and the right image is coded with disparity estimation in the enhancement layer. Since it is possible to calculate rough motion of the right image sequence with disparity and motion of the left image sequence, motion compensation of the enhancement layer is performed without motion estimation. To apply this mathod to MVP codec., the prediction mode of base layer and enhancement layer is restricted, and B picture mode in the base layer is removed. Since the proposed codec. does not perform motion estimation in the enhancement layer encoding and prediction mode of base layer is restricted, it's structure is simple and reduces the encoding time. We compared the SNR of encoded image with three different structured codec., and the experimental results show suggested codec. have comparable result.

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