• Journal of the Institute of Electronics and Information Engineers
• /
• v.50 no.1
• /
• pp.86-96
• /
• 2013

Unified Speech and Audio Coding (USAC) is the speech/audio codec with the best quality, approved on Final Draft International Standard (FDIS) at MPEG meeting in 2011. Since MPEG conventionally standardizes only the decoder, it is not easy to study on the encoder technologies. Furthermore, Reference Model(RM) shows extremely poor performance. To solve these problems, the open source project(JAME) proposes the methods to make the improved performance of main encoder technologies in USAC. Especially, this paper introduces the encoder modules: the signal classifier for selective operation between two coders, the psychoacoustic model in frequency domain, and window transition technology. Finally, the results of verification test for FDIS and the performance of Common Encoder are appended.

• The KIPS Transactions:PartA
• /
• v.13A no.4 s.101
• /
• pp.275-288
• /
• 2006

Rapid design space exploration is crucial to customizing embedded system design for exploiting the application behavior. As the time-to-market becomes a key concern of the design, the approach based on an application specific instruction-set processor (ASIP) is considered more seriously as one alternative design methodology. In this approach, the instruction set architecture (ISA) for a target processor is frequently modified to best fit the application with regard to code size and speed. Two goals of this paper is to introduce our new retargetable compiler and how it has been used in ASIP-based design space exploration for a popular digital signal processing (DSP) application. Newly developed retargetable compiler provides not only the functionality of previous retargetable compilers but also visualizes the features of the application program and profiles it so that it can help architecture designers and application programmers to insert new application specific instructions into target architecture for performance increase. Given an initial RISC-style ISA for the target processor, we characterized the application code and incrementally updated the ISA with more application specific instructions to give the compiler a better chance to optimize assembly code for the application. We get 32% performance increase and 20% program size reduction using 6 audio codec specific instructions from retargetable compiler. Our experimental results manifest a glimpse of evidence that a higgly retargetable compiler is essential to rapidly prototype a new ASIP for a specific application.

• 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.

• 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.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.47 no.3
• /
• pp.87-95
• /
• 2010

Portable multi-media products which can service the highest audio-quality by using lossless audio codec has been released and the international lossless codecs, MPEG-4 audio lossless coding(ALS) and MPEG-4 scalable lossless coding(SLS), were standardized by MPEG in 2006. The simple profile of MPEG-4 ALS, it supports up to stereo, was defined by MPEG in 2009. The lossless audio codec should have low-complexity in stereo to be widely used in portable multi-media products. But the previous researches of MPEG-4 ALS have focused on an improvement of compression ratio, a complexity reduction in multi-channels coding, and a selection of linear prediction coefficients(LPCs) order. In this paper, the complexity and compression ratio of MPEG-4 ALS encoder is analyzed in simple profile of MPEG-4 ALS, the method to reduce a complexity of MPEG-4 ALS encoder is proposed. Based on an analysis of complexity of MPEG-4 ALS encoder, the complexity of short-term prediction filter of MPEG-4 ALS encoder is reduced by using the low-complexity filter that is proposed in previous research to reduce the complexity of MPEG-4 ALS decoder. Also, we propose a joint coding decision method, it reduces the complexity and keeps the compression ratio of MPEG-4 ALS encoder. In proposed method, the operation of joint coding is decided based on the relation between cross-correlation of residual and compression ratio of joint coding. The performance of MPEG-4 ALS encoder that has the method and low-complexity filter is evaluated by using the MPEG-4 ALS conformance test file and normal music files. The complexity of MPEG-4 ALS encoder is reduced by about 24% by comparing with MPEG-4 ALS reference encoder, while the compression ratio by the proposed method is comparable to MPEG-4 ALS reference encoder.

• Proceedings of the IEEK Conference
• /
• 2006.06a
• /
• pp.257-258
• /
• 2006

In this paper, we address several issues in the real time implementation of MPEG-1 Layer II decoder on a fixed-point digital signal processor (DSP), especially TMS320C6416. There is a trade-off between processing speed and the size of program/data memory for the optimal implementation. In a view of the speed optimization, we first convert the floating point operations into fixed point ones with little degradation in audio quality, and then the look-up tables used for the inverse quantization of the audio codec are forced to be located into the internal memory of the DSP. And then, window functions and filter coefficients in the decoder are precalculated and stored as constant, which makes the decoder faster even larger memory size is required. It is shown from the real-time experiments that the fixed-point implementation enables us to make the decoder with a sampling rate of 48 kHz operate with 3 times faster than real-time on TMS320C6416 at a clock rate of 600 MHz.

• Proceedings of the IEEK Conference
• /
• 2006.06a
• /
• pp.507-508
• /
• 2006

High performance delta-sigma modulator is developed for audio-codec applications(i.e.. 16-bit resolution at a 20kHz signal bandwidth). The modulator is realized with fully-differential switched capacitor integrators. All stages employ a single-stage folded-cascode amplifier. The presented delta-sigma modulator when clocked at 3.2MHz achieves 85.2dB peak-SNDR and 94.8dB SNR. This modulator is designed in a SAMSUNG $0.18{\mu}m$ CMOS process. Finally, this paper shows the test setup and FFT result gained from delta-sigma modulator chip designed for audio applications.

• Journal of Broadcast Engineering
• /
• v.22 no.6
• /
• pp.829-849
• /
• 2017

Due to the large-scale TV display, the convergence of broadcasting and broadband, and the advancement of signal compression and transmission technology, terrestrial digital broadcasting has evolved into UHD broadcasting capable of providing simultaneous broadcasting of fixed UHD and mobile HD. The Korean standard for terrestrial UHDTV broadcasting is based on ATSC 3.0, the broadcasting standard of North America. The terrestrial UHDTV broadcasting standard chose that as a new AV codec standard, HEVC video codec which can compress with higher efficiency compared to AVC, and MPEG-H 3D audio codec for realistic audio. Also, DASH and MMT are adopted as transmission format instead of MPEG-2 TS to support broadband as well as broadcasting network, and in order to provide 4K UHD/mobile HD service simultaneously ROUTE multiplexing technology is applied. In this paper, we propose an audio/video encoder, which is required to provide HDR/WCG supported high quality video service, 10.2 channel/4 object supporting stereo sound service, fixed UHD and mobile HD simultaneous broadcasting service based on ATSC3.0, also we implemented the ATSC 3.0 LDM system for ROUTE/DASH packager, multiplexing system and physical layer transmission/reception, and verified the service ability by applying it to real time broadcast environment.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 1999.11b
• /
• pp.261-266
• /
• 1999

본 논문에서는 인터넷 폰이나 디지털 이동통신에서와 같이 낮은 비트율이 요구되는 응용분야에서 사용될 수 있는 HVXC 부호화 및 복호화 알고리즘을 TMS320C6701 160MHz DSP를 사용하여 실시간 동작을 구현한 내용을 기술한다. 사용한 최적화 방법으로는 기본적으로 연산 시간이 많이 소요되는 함수 루틴에 대한 C 언어레벨의 최적화 및 어셈블리어 레벨의 최적화를 수행하였고, TMS320C6701 DSP 내부 프로그램 메모리를 프로그램 캐쉬로 사용하였다. 또한, 계산량이 많은 부분과 테이블 참조가 필요한 연산을DSP의 내부 데이터 메모리 영역에서 수행하여 소요시간을 단축하였으며, 음성신호 및 비트스트림의 입출력에는 background DMA(direct memory access) 방식을 이용하였다. 이와 같은 최적화결과 2kbps 및 4kbps의 비트율에서 압축 및 복원을 실시간으로 수행할 수 있다.

• Proceedings of the KIEE Conference
• /
• 2008.10b
• /
• pp.304-305
• /
• 2008

DVD/BD 및 HDTV의 보급으로 인해 다수의 오디오 컨텐츠들이 멀티채널(5.1채널 이상) 형식으로 제작되고 있다. 오디오 정보를 담고 있는 물리적인 채널의 수가 증가하면, 이에 따라 정보량도 선형적으로 증가하게 된다. 선형적으로 증가된 정보를 기존의 오디오 코덱을 이용해 큰 압축할 경우, 압축에 필요한 비트레이트의 선형적인 증가를 피할 수 없다. 최근 채널 수 증가로 야기되는 비트레이트의 증가를 최소화하고 효율적으로 멀티채널 오디오 신호를 압축할 수 있는 방법으로 MPEG surround와 같은 파라메트릭 멀티채널 오디오 코딩 방식이 제안되었다. 파라메트릭 멀티채널 오디오 코딩 방식의 경우, 멀티채널 오디오 신호를 채널 수가 감소된 다운믹스 신호와 다운믹스 신호로부터 다시 멀티채널 오디오 업믹스 하는데 필요한 파라미터로 표현하는 방식이다. 따라서 다운믹스 방식 및 업믹스에 필요한 파라미터에 따라 업믹스된 멀티채널 오디오 신호의 품질이 달라진다. 본 논문에서는 MPEG surround에서 사용하고 있는 기존의 ITU-R 다운믹스 방식의 문제점을 실제 멀티채널 오디오 신호의 사례를 통해 제시하고 이 문제점을 해결하기 위한 새로운 다운믹스 방식과 파라미터를 제안하고자 한다.