• The Journal of the Acoustical Society of Korea
• /
• v.32 no.2
• /
• pp.157-166
• /
• 2013

This paper proposes a non-intrusive speech quality estimation method considering the effects of packet loss to perceptual quality. Packet loss is a major reason of quality degradation in a packet based speech communications network, whose effects are different according to the input speech characteristics or the performance of the embedded packet loss concealment (PLC) algorithm. For the quality estimation system that involves packet loss effects, we first observe the packet loss of G.729 codec which is one of narrowband codec in VoIP system. In order to quantify the lost packet affects, we design a classification algorithm only using speech parameters of G.729 decoder. Then, the degradation values of each class are iteratively selected that maximizes the correlation with the degradation PESQ-LQ scores, and total quality degradation is modeled by the weighted sum. From analyzing the correlation measures, we obtained correlation values of 0.8950 for the intrusive model and 0.8911 for the non-intrusive method.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.15 no.1
• /
• pp.58-70
• /
• 1990

This paper describes a VLSI implementation of real-time two dimensional DCT processor for the subprimary rate video codec system. The proposed architecture exploits the parallelism and concurrency of the distributes architecture for vector inner product operation of DCT and meets the CCITT performance requirements of video codec for full CSIF 30 frames/sec. It is also shown that this architecture satisfies all the CCITT IDCT accuracy specification by simulating the suggested architecture in bit level. The efficient VLSI disign methodology to design suggested architecture is considered and the module generator oriented design environments are constructed based on SUN 3/150C workstation. Using the constructed design environments. the suggensted architecture have been designed by double metal 2micron CMOS technology. The chip area fo designed 8x8 2-D DA-DCT (Distributed Arithmetic DCT) processor is about 3.9mmx4.8mm.

• The Journal of the Acoustical Society of Korea
• /
• v.36 no.1
• /
• pp.70-77
• /
• 2017

Bandwidth extension is a technique to improve speech quality, intelligibility and naturalness, extending from the 300 ~ 3,400 Hz narrowband speech to the 50 ~ 7,000 Hz wideband speech. In this paper, an Artificial Bandwidth Extension (ABE) module embedded in the Opus audio decoder is designed using the information of narrowband speech to reduce the computational complexity of LPC (Linear Prediction Coding) and LSF (Line Spectral Frequencies) analysis and the algorithm delay of the ABE module. We proposed a spectral envelope extension method using DBN (Deep Belief Network), one of deep learning techniques, and the proposed scheme produces better extended spectrum than the traditional codebook mapping method.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.31 no.3A
• /
• pp.337-344
• /
• 2006

In this paper, we propose a multi-view video codec for 3DTV system. The proposed algorithm is not only to reduce the temporal and spatial redundancy but also to reduce the redundancy among each view. With these results, we can improve the coding efficiency for multi-view video sequences. In order to reduce the redundancy of each view more efficiently, we define the assembled image(AI) that is generated by the global disparity compensation of each view. In addition, the proposed algorithm is based on MPEG-2 structure so that we can easily implement 3DTV system without changing the conventional 2D digital TV system. Experimental results show that the proposed algorithm performs very well. It also performs better than MPEG-2 simulcast coding method. The newly proposed codec also supports the view scalability, accurate temporal synchronization among multiple views and random access capability in view dimension.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.15 no.2
• /
• pp.439-446
• /
• 2011

In this paper, we proposed and implemented a dedicated hardware for Huffman coding which is a method of entropy coding to use compressing multimedia data with video coding. The proposed Huffman codec consists Huffman encoder and decoder. The Huffman encoder converts symbols to Huffman codes using look-up table. The Huffman code which has a variable length is packetized to a data format with 32 bits in data packeting block and then sequentially output in unit of a frame. The Huffman decoder converts serial bitstream to original symbols without buffering using FSM(finite state machine) which has a tree structure. The proposed hardware has a flexible operational property to program encoding and decoding hardware, so it can operate various Huffman coding. The implemented hardware was implemented in Cyclone III FPGA of Altera Inc., and it uses 3725 LUTs in the operational frequency of 365MHz

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.19 no.2
• /
• pp.386-392
• /
• 2015

Data in a image compression codec are processed with a specific regular block size. The processing order of block sized data is changed in specific function blocks and the data is packed in memory and read by a new sequence. To maintain a regular throughput rate, double buffering is normally used that interleaving two block sized memory to do concurrent read and write operations. Single buffering using only one block sized memory can be adopted to the simple data reordering, but when a complicate reordering occurs, irregular address changes prohibit from implementing adequate address generating for single buffering. This paper shows that there is a predictable and recurring regularity of changing address access orders within a finite updating counts and suggests an effective method to implement. The data reordering function using suggested idea is designed with HDL and implemented with TSMC 0.18 CMOS process library. In various scan blocks, it shows more than 40% size reduction compared with a conventional method.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.13 no.2
• /
• pp.962-977
• /
• 2019

The human voice is a convenient method of information transfer between different objects such as between men, men and machine, between machines. The development of information and communication technology, the voice has been able to transfer farther than before. The way to communicate, it is to convert the voice to another form, transmit it, and then reconvert it back to sound. In such a communication process, a vocoder is a method of converting and re-converting a voice and sound. The CELP (Code-Excited Linear Prediction) type vocoder, one of the voice codecs, is adapted as a standard codec since it provides high quality sound even though its transmission speed is relatively low. The EVRC (Enhanced Variable Rate CODEC) and QCELP (Qualcomm Code-Excited Linear Prediction), variable bit rate vocoders, are used for mobile phones in 3G environment. For the real-time implementation of a vocoder, the reduction of sound quality is a typical problem. To improve the sound quality, that is important to know the size and shape of noise. In the existing sound quality improvement method, the voice activated is detected or used, or statistical methods are used by the large mount of data. However, there is a disadvantage in that no noise can be detected, when there is a continuous signal or when a change in noise is large.This paper focused on finding a better way to decrease the reduction of sound quality in lower bit transmission environments. Based on simulation results, this study proposed a preprocessor application that estimates the SNR (Signal to Noise Ratio) using the spectral SNR estimation method. The SNR estimation method adopted the IMBE (Improved Multi-Band Excitation) instead of using the SNR, which is a continuous speech signal. Finally, this application improves the quality of the vocoder by enhancing sound quality adaptively.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2020.11a
• /
• pp.377-379
• /
• 2020

The hologram data is having a dependence on the pixel pitch of the SLM (spatial light modulator) and the wavelength of light, and the quality of the digital hologram is proportional to the unit pixel pitch and the total resolution. In addition, since each pixel has a complex value, the amount of data in the digital hologram also increases exponentially, and the size is bound to be very large. Therefore, in order to efficiently handle digital hologram files, it is essential to reduce the file size through a codec and store it. Recently, research on enhancing image quality damaged by the codec is actively underway. In this paper, the hologram image of JPEG Pleno, which is the standard hologram data, was used, and the image quality damage that occurs whenthe holographic image is encoded and decoded through the JPEG2000, AVC, and HEVC codec is enhanced with a deep learning network to find out whether the image quality can be improved. we also compare and quantitatively find out the degree of improvement in image quality.

• The Journal of the Acoustical Society of Korea
• /
• v.27 no.1
• /
• pp.18-25
• /
• 2008

In this paper, new lossless coding method of spectral coefficients for audio codec is proposed. Conventional lossless coder uses Huffman coding utilizing the statistical characteristics of spectral coefficients, but does not provide the high coding efficiency due to its simple structure. To solve this limitation, new lossless coding scheme with better performance is proposed that consists of bit-plane transform and run-length coding. In the proposed scheme, the spectral coefficients are first transformed by bit-plane into 1-D bit-stream with better correlative properties, which is then coded intorun-length and is finally Huffman coded. In addition, the coding performance is further increased by applying the proposed bit-plane coding selectively to each group, after the entire frequency is divided into 3 groups. The performance of proposed coding scheme is measured in terms of theoretical number of bits based on the entropy, and shows at most 6% enhancement compared to that of conventional lossless coder used in AAC audio codec.

• The Journal of the Acoustical Society of Korea
• /
• v.26 no.6
• /
• pp.301-305
• /
• 2007

In this paper, we propose a simple algorithm for detecting underwater transient signals on the fact that the frequency range of underwater transient signals is similar to audio frequency. For this, we use a preprocessing module of EVRC speech codec that is the standard speech codec of the mobile communications. If a signal is entered into EVRC noise suppression module, we can get some parameters such as the update flag, the energy of each channel, the noise suppressed signal, the energy of input signal, the energy of background noise, and the energy of enhanced signal. Therefore the energy of the enhanced signal that is normalized with the energy of the background noise is compared with the pre-defined detection threshold, and then we can detect the transient signal. And the detection threshold is updated using the previous value in the noisy period. The experimental result shows that the proposed algorithm has $0{\sim}4% error rate in the AWGN or the colored noise environment.