• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.8C
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• pp.822-829
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• 2009

This paper proposes a new method of improving the quality of High Efficiency Advanced Audio Coding (HE-AAC). HE-AAC encodes input source by allocating bits for each scalefactor bands appropriately according to human ear's psychoacoustic property. As a result, insufficient bits are assigned to the bands which have relatively low energy. This imbalance between different energy bands can cause decreasing of sound quality like musical noise. In the proposed system, a Linear Prediction (LP) module is combined with HE-AAC as a pre-processor to improve sound quality by even bits distribution. To apply accurate human being's psychoacoustic property, the psychoacoustic model uses Fast Fourier Transform (FFT) spectrum of original input signal to make masking threshold. In its implementation, masking threshold of psychoacoustic model is normalized using the LP spectral envelope in prior to quantization of the LP residual. Experimental result shows that, the proposed algorithm allocates bits appropriately for insufficient bits condition and improves the performance of HE-AAC.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2008.11a
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• pp.251-253
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• 2008

최근 유럽에서 DVB-T HDTV 방송 표준이 정하지면서 오디오 포맷으로 HE-AAC가 채택되었다. HE-AAC는 압축효율은 높지만 연산량이 높아 낮은 성능의 DSP에서 수행하기에는 어려움이 있다. DVB-T에서는 5.1채널을 사용하고 있어 더욱더 많은 연산을 필요로 한다. 본 논문은 ISO/DEC 14496-3 MPEG4 HE(High Efficiency)-AAC의 Level4에 해당하는 Multichannel Decoder를 최적화하여 구현하고. 가장 많은 연산을 필요로 하는 Synthesis Filter Bank에 제안된 알고리즘을 적용하여 연산량을 줄였고 대부분의 연산부를 어셈블리로 코드 최적화를 하여 작은 성능의 DSP를 사용하여 실시간 Multichannel HE-AAC Audio Decoder의 구현이 가능하게 하였다. DVB-T 오디오 시스템에 필수로 필요한 Audio Description, Dynamic Range Control, Downmix 등을 함께 구현하여 실제 수신기에 사용이 가능하도록 하였다. DSP는 Samsung의 CalmRISC16 + MAC24 core 를 사용하였다.

• The Journal of the Acoustical Society of Korea
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• v.27 no.2
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• pp.66-74
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• 2008

In this paper, we propose new techniques that can improve the quality of AAC and SBR encoders comprised in low bitrate HE-AAC. To reduce the pre-echo artifacts often occurring for transient blocks in AAC, we propose an extended Temporal Noise Shaping (sTNS) in which the frequency range is selectively extended down to the low-frequency region. Also, for he high-frequency region being coded by SBR encoder, tones are identified through a sinusoidal modeling and their frequencies are adjusted within the QMF band in order to reduce the noise floor due to aliasing. Spectrograms of the decoded signals were compared and listening tests were conducted to evaluate the proposed algorithm. Results confirmed the effectiveness of the proposed algorithm.

• Journal of Broadcast Engineering
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• v.21 no.3
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• pp.357-368
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• 2016

In DAB+, HE-AAC v2 codec is applied for the fixed rate transmission of audio stream. In case that xHE-AAC codec including USAC, a more efficiency is expected when the variable frame is used in a given same bandwidth compared to the fixed frame transmission. For this to be realized, audio streams need to be multiplexed in a sub-channel before transmission, then a method is required to identify the border of each audio frames. In this paper, the toggled sync byte and additional identification field being sequentially placed between AU borders are proposed in order to deal with the AU border identification. In addition, the Reed-Solomon based error correction code which is compliant to DAB+ is proposed.

• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.714-721
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• 2011

MPEG audio uses the masking effect, high frequency component synthesis based on spectral band replication, and channel expansion based on parametric stereo for efficient compression of audio signals. In this paper, we present an overview of the state-of-the-art channel expansion technology in MPEG audio. We also present technical overviews and application examples to broadcasting services for HE-AAC v.2, MPEG Surround, spatial audio object coding (SAOC), and unified speech and audio coding (USAC) which are MPEG audio codecs based on the channel expansion technology.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2006.07a
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• pp.429-429
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• 2006

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.589-598
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• 2009

As mobile devices become multi-functional, and converge into a single platform, there is a strong need for a codec that is able to provide consistent quality for speech and music content MPEG-D USAC standardization activities started at the 82nd MPEG meeting with a CfP and approved WD3 at the 88th MPEG meeting. MPEG-D USAC is converged technology of AMR-WB+ and HE-AAC V2. Specifically, USAC utilizes three core codecs (AAC ACELP and TCX) for low frequency regions, SBR for high frequency regions and the MPEG Surround tool for stereo information. USAC can provide consistent sound quality for both speech and music content and can be applied to various applications such as multi-media download to mobile device Digital radio Mobile TV and audio books.

• The Journal of the Acoustical Society of Korea
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• v.25 no.4E
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• pp.137-141
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• 2006

This paper presents HE-AAC decoder optimization on TMS320C55x fixed-point DSP core using a DSP-C like FFR code, which provides fast and flexible porting to a DSP core. Our optimization efforts are focused on methodologies that include general optimization methods of FFR code suitable for general DSP or RISC platform in high-level language and software optimization methods in assembly language level. The implementation result requires 48 MIPS and 135 Kbytes memory space to decode 48 Kbps stereo using real Korean S-DMB data.

• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.693-704
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• 2011

As mobile devices become multi-functional, and converge into a single platform, there is a strong need for a codec that is able to provide consistent quality for speech and music contents. MPEG-D USAC standardization activities started at the 82nd MPEG meeting with a CfP and approved Study on DIS at the 96th MPEG meeting. MPEG-D USAC is converged technology of AMR-WB+ and HE-AAC V2. Specifically, USAC utilizes three core codecs (AAC, ACELP, and TCX) for low frequency regions, SBR for high frequency regions, the MPEG Surround for stereo information, and window transition technology for smoothing transition between various core coder. USAC can provide consistent sound quality for both speech and music contents and can be applied to various applications such as multi-media download to mobile devices, digital radio, mobile TV and audio books.

• The Journal of the Acoustical Society of Korea
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• v.28 no.1E
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• pp.29-34
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• 2009

MPEG4 High Efficiency Advanced Audio Coding (HE-AAC) and MPEG Surround are one of the most efficient combinations for low bit rate multi-channel audio coding. Based on the fact that these two codecs have identical quadrature mirror filter (QMF) analysis and synthesis structures, we propose a dual-domain connection scheme for the codecs. Specifically two time-domain connection methods are analyzed and compared to the QMF subband-domain connection method. Experimental results show that both the time-domain connection methods cause no subjective sound quality degradation compared to the QMF subband-domain connection method, which verifies that one can select either of them depending on application scenarios.