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检索条件"主题词=Audio coding"

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Aliasing Reduction for Modified Discrete Cosine Transform Domain Filtering and its Application to Speech Enhancement

Aliasing Reduction for Modified Discrete Cosine Transform Do...

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Fabian Kuech Bernd Edler Fraunhofer Institute of Integrated Circuits and Systems Erlangen Germany Laboratorium für Informationstechnologie Leibniz Universität Hannover Hanover Germany

Efficient combinations of coding and manipulation of audio signals in the spectral domain are often desirable in communication systems. The modified discrete cosine transform (MDCT) represents a popular spectral transform in audio coding as it leads to compact signal representations. However, as the MDCT corresponds to a critically sampled filter bank, it is in general not appropriate to directly apply it to filtering tasks. In this paper we present a method to compensate for aliasing terms that arise from such direct MDCT domain filtering. The discussion is thereby based on a rigorous matrix representation of critically sampled filter banks which also leads to corresponding efficient realizations. As an application showcase, noise reduction for MDCT based speech coding is considered in simulations.

关键词： Discrete cosine transforms Filtering Speech enhancement Filter bank Noise reduction audio coding Discrete Fourier transforms Time domain analysis Discrete transforms Signal representations

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Adaptive Bit-Plane Scanning for Scalable audio

Adaptive Bit-Plane Scanning for Scalable Audio

引用

IEEE International Conference on Multimedia and Expo (ICME)

作者： Te Li Susanto Rahardja Soo Ngee Koh Institute for Infocomm Research Singapore Nanyang Technological University Singapore

MPEG-4 scalable lossless (SLS) coding is a unified solution for demands in high compression perceptual audio and high quality lossless audio. It provides a fine-grain scalable extension to the MPEG-4 advanced audio coding (AAC) perceptual audio coder up to fully lossless reconstruction. It is observed that the quality of SLS coded audio is still far from optimal when the core bitrate is low. With this observation, an adaptive bit-plane scanning (ABPS) method is proposed in this paper. In ABPS, the bit-plane scanning order is adaptive to the energy distribution of the signal to be coded. The results show that with ABPS, the perceptual quality of the audio under low core bitrate scenario is significantly improved without any extra payload.

关键词： Laser sintering Bit rate audio coding Frequency conversion MPEG 4 Standard Scalability Tellurium Image reconstruction Payloads Computer networks

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Direct MDCT Domain Psychoacoustic Modeling

Direct MDCT Domain Psychoacoustic Modeling

引用

IEEE International Symposium on Signal Processing and Information Technology (ISSPIT)

作者： K Suresh T.V. Sreenivas Department of Electrical Communication Engineering Indian Institute of Science Bangalore India

We extend the recently proposed spectral integration based psychoacoustic model for sinusoidal distortions to the MDCT domain. The estimated masking threshold additionally depends on the sub-band spectral flatness measure of the signal which accounts for the non- sinusoidal distortion introduced by masking. The expressions for masking threshold are derived and the validity of the proposed model is established through perceptual transparency test of audio clips. Test results indicate that we do achieve transparent quality reconstruction with the new model. Performance of the model is compared with MPEG psychoacoustic models with respect to the estimated perceptual entropy (PE). The results show that the proposed model predicts a lower PE than other models.

关键词： Psychology Psychoacoustic models Frequency domain analysis Masking threshold Humans Signal processing Distortion Transform coding Auditory system audio coding

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A Leading Sign Grouping with Direct Table Lookup Approach for AAC Huffman Decoding

A Leading Sign Grouping with Direct Table Lookup Approach fo...

引用

IEEE International Symposium on Multimedia Workshops (ISMW)

作者： Chun-Nan Liu Tsung-Han Tsai Shau-Yin Tseng Department of Electrical Engineering National Central University Taiwan SoC Technology Center Industrial Technology and Research Institute Hsinchu Taiwan

ISBN: (纸本)9780769530840;0769530842

This paper presents a high speed search and memory efficient Huffman decoding for MPEG AAC audio decoder. First, the leading sign bit characteristic in AAC Huffman codewords is utilized. Based on the leading sign grouping, a two-level table lookup approach is introduced. It can branch to the correct group by first table and decode the symbols by second table. Two optimization approaches are developed, one is to reduce the branch cycles and the other is to combine direct table lookup approach. Simulation results show that these two optimizations can totally save about 18.25% cycles further comparing to the non-optimized version and about 9x speed up comparing to the linear search approach. Besides, the codeword packing scheme is used that the 12 Huffman codebooks only consume 3.6 Kbytes.

关键词： Table lookup Decoding Digital signal processing Computer architecture VLIW Parallel processing Binary trees audio coding Signal processing algorithms Conferences

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Analysis and Application of Perceptual Weighting for Avs-M audio Coder

Analysis and Application of Perceptual Weighting for Avs-M A...

引用

International Conference on Wireless Communications, Networking and Mobile Computing (WiCom)

作者： Yuhong Yang Ruimin Hu Yong Zhang Wei Zhang The Key Laboratory of Multimedia and Network Communications Engineering Wuhan Univ. Hubei China

audio video coding standard (AVS) is the latest audio and video coding standard of China. AVS Part 7 (also known as AVS-M) audio coding tools target mobility applications where perceptual weighting is of great importance for subjective performance. This paper first briefly introduces the general concept of perceptual weighting. Then a candidate perceptual weighting scheme is proposed and testified on AVS-M audio coder. Further optimization on the candidate is illustrated thereafter. Subjective results of the optimized scheme provide hybrid audio quality comparable to the 3 GPP AMR-WB+.

关键词： audio coding Distortion measurement Bit rate Video coding Noise shaping Frequency Code standards Communication standards Testing Speech coding

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Spatial Cue Based Sound Scene Control for MPEG Surround

Spatial Cue Based Sound Scene Control for MPEG Surround

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IEEE International Conference on Multimedia and Expo (ICME)

作者： Seungkwon Beack Jeongil Seo Taejin Lee Dae-young Jang Electronics and Telecommunications Research Institute South Korea

Spatial audio coding scheme is advanced technology in the sense of saving bitrate without any scarifications of sound quality. In this paper, the method to utilizing the spatial cues is present. This utilization is for supporting a new useful functionality, i.e., flexible sound scene control. The function of sound scene control may result in providing user preferable audio service by interaction and be utilized in upcoming applications such as multi-view video service and virtual reality applications. It will be shown that even with a little addition of computational operations; this useful functionality will be achievable maintaining the corresponding sound quality.

关键词： Layout Cross layer design MONOS devices Decoding audio coding Signal processing Auditory system Bit rate Humans Amplitude estimation

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Novel Non-linear Inverse Quantization Algorithm and its Architecture for Digital audio Codecs

Novel Non-linear Inverse Quantization Algorithm and its Arch...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Han Jae Hee Myung Hoon Sunwoo Jong Ha Moon School of Electrical and Computer Engineering Ajou University South Korea SAT Gr. Digital TV Laboratory LG Electronics Inc. Seoul South Korea

This paper proposes a novel inverse-quantization (IQ) algorithm and specialized DSP instructions for IQ algorithms. Non-linear inverse quantization is essential in both MPEG-1 Layer-3 and MPEG-2/4 advanced audio coding (AAC). The proposed algorithm is optimized for the non-linear inverse quantization. The proposed algorithm has moderate computational complexity and can achieve the minimum average error. Performance comparisons show a significant improvement compared with existing algorithms. Furthermore, we propose the specialized DSP instructions and their architecture that can efficiently support the proposed IQ algorithm and other IQ algorithms. The audio-specific instruction-set processor that we have previously implemented employs these instructions and their hardware.

关键词： Quantization Codecs Digital audio players Digital signal processing Table lookup ISO standards IEC standards Computer architecture audio coding Approximation algorithms

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A Low-Delay audio Coder with Constrained-Entropy Quantization

A Low-Delay Audio Coder with Constrained-Entropy Quantizatio...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Minyue Li W. Bastiaan Kleijn ACCESS Linnaeus Center Electrical Engineering KTH Royal Institute of Technology Stockholm Sweden

We present a low-delay, constrained-entropy, backward adaptive, linear-predictive audio coder with low computational complexity. In contrast to most practical linear-predictive coders, the coder facilitates the exploitation of reverse waterfilling. The coder uses time-invariant quantization step size and constrained-entropy coding, thus eliminating the convergence problems of backward adaptation near signal transitions. Yet rate variations are kept small by the usage of a mixture model density for the signal. The mixture model has the backward adapted model and a second model as components and the component probability is transmitted. Experimental results confirm the advantages of the coder structure and show that the coder provides good overall performance.

关键词： Quantization audio coding Transform coding Predictive coding Delay effects Echo cancellers Hearing aids Delay estimation Conferences Acoustic signal processing

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Rate Distribution Between Model and Signal

Rate Distribution Between Model and Signal

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： W. Bastiaan Kleijn Alexey Ozerov ACCESS Linnaeus Center Electrical Engineering KTH Royal Institute of Technology Stockholm Sweden

Knowledge of a statistical model of the signal can be used to increase coding efficiency. A common approach is to use a fixed model structure with parameters that adapt to the signal. The model parameters and a signal representation that depends on the model are encoded. We show that, if the signal is divided into segments of a particular duration, and the model structure is fixed, then the optimal bit allocation for the model parameters does not vary with the overall rate. We discuss in detail the parameter rate for the autoregressive (AR) model. Our approach shows that the square error criterion in the signal domain is consistent with the commonly used root mean square log spectral error for the model parameters. Without using perceptual knowledge, we obtain a rate allocation for the model that is consistent with what is commonly used. This model rate is independent of overall coding rate. We provide experimental results for the application of the autoregressive model to speech that confirm the theory.

关键词： Bit rate Speech coding Signal processing Encoding Statistics Source coding audio coding Conferences Acoustic signal processing Acoustic applications

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Modeling Spot Microphone Signals using the Sinusoidal Plus Noise Approach

Modeling Spot Microphone Signals using the Sinusoidal Plus N...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Christos Tzagkarakis Athanasios Mouchtaris Panagiotis Tsakalides Department of Computer Science Institute of Computer Science FORTH-ICS University of Crete Greece

This paper focuses on high-fidelity multichannel audio coding based on an enhanced adaptation of the well-known sinusoidal plus noise model (SNM). Sinusoids cannot be used per se for high-quality audio modeling because they do not represent all the audible information of a recording. The noise part has also to be treated to avoid an artificial sounding resynthesis of the audio signal. Generally, the encoding process needs much higher bitrates for the noise part than the sinusoidal one. Our objective is to encode spot microphone signals using the SNM, by taking advantage of the interchannel similarities to achieve low bitrates. We demonstrate that for a given multichannel audio recording, the noise part for each spot microphone signal (before the mixing stage) can be obtained by using its noise envelope to transform the noise part of just one of the signals (the so-called "reference signal", which is fully encoded).

关键词： Microphones Acoustic noise audio recording Bit rate audio coding Computer science Frequency Image coding Decoding Multiple signal classification

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