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检索条件"主题词=Audio coding"

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A SPATIAL PRIORITY BASED SCALABLE audio coding

A SPATIAL PRIORITY BASED SCALABLE AUDIO CODING

引用

IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Li Gao Ruimin Hu Yuhong Yang National Engineering Research Center for Multimedia Software Computer School of Wuhan University China

ISBN: (纸本)9781479928941

A spatial priority scheme for scalable audio coding is presented in this paper. To improve the coding quality of important sounds with high attention, especially the moving sound, spatial information is introduced to assign the priorities of frequency subbands. Spatial cues and distance features are extracted in frequency subbands to represent the sound with fast changing direction and distance. coding priorities are assigned to different frequency subbands according to the energy and spatial information. With trivial added side information and complexity, experimental results show that the perceptual quality is improved especially for the sound with high attention, especially the moving sound in scalable audio coding.

关键词： audio coding SOUND Vibration Hearing Subbands audio Scalability

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Lossless audio coding using Burrows-Wheeler Transform and move-to-front coding

Lossless audio coding using Burrows-Wheeler Transform and mo...

引用

International Conference on Computer Engineering and Systems

作者： Hend Ali Elsayed Masoud Alghoniemy Department of Electrical Engineering Alexandria University Alexandria Egypt

ISBN: (纸本)1424413656;9781424413652

In this paper we present a new lossless audio coder using the Burrows-Wheeler transform (BWT). audio signals under consideration, which are assumed to be of integer values, are first permuted using the BWT to get the transform coefficients; and then the resulting coefficients are compressed using a combination of the move-to-front coding (MTF) and an entropy coder. Both Arithmetic and Huffman coding have been used for entropy coding. Simulations show that the proposed coder outperforms other lossless coders, namely, the integer discrete cosine transform (intDCT) and the integer wavelet transform (intWT).

关键词： audio coding Discrete transforms Image reconstruction Discrete wavelet transforms Sorting Entropy coding Streaming media Arithmetic Huffman coding Discrete cosine transforms

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Subband audio coding with synthesis filters minimizing a perceptual distortion

Subband audio coding with synthesis filters minimizing a per...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： K. Gosse F. Moreau de Saint-Martin X. Durot P. Duhamel J.B. Rault Centre de Recherche Motorola USA Department Signal ENST Paris France Rue du Clos Courtel CCETT Cesson-Sevigne France

The design of filter banks for source coding purposes classically relies on the perfect reconstruction (PR) property. However, several studies have shown that taking the quantization noise into account in the design could yield a noticeable reduction of the mean square reconstruction error. The purpose of this study is to show that perceptual improvement can also be obtained in the particular audio coding context by relaxing the PR constraint. In this context, the mean square error is not relevant any more, and we define a new perceptual distortion criterion, making use of a simplified ear model, the MPE (mean perceptual error). Then, synthesis filters are optimized so as to minimize this MPE. Finally, this MMPE (minimum MPE) filter bank is included in an audio coding scheme. Compared to the corresponding PR filter bank-based scheme by the means of POM (perceptual objective measure), they show an improved audio quality.

关键词： audio coding Filter bank Wiener filter Quantization Signal synthesis Auditory system Mean square error methods Ear audio compression Distortion measurement

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Attack restoration in low bit-rate audio coding, using an algebraic detector for attack localization

Attack restoration in low bit-rate audio coding, using an al...

引用

International Symposium on I/V Communications and Mobile Network (ISVC)

作者： Imen Samaali Monia Turki-Hadj Alouane Gael Mané Signals and Systems Research Unit National Engineering School of Tunis (ENIT) Tunis Tunisia LIPADE University Paris Descartes Paris France Signals and Systems Research Unit(U2S) National Engineering School of Tunis (ENIT) Tunis Tunisia

This paper deals with pre-echo reduction in low bit-rate audio compression. proposed an attack restoration method based on the correction of the temporal envelop of the decoded signal. A small set of coefficients were then transmitted through a limited bit-rate auxiliary channel. However, the transmission of the transient position computed on the original audio signal was required. In this paper, we deployed a new method of attack localization based on differential algebraic, which guaranties a successful detection on the decoded audio signal. The algebraic method has also a reduced complexity compared to the index stationary detector used in. The new proposed approach is evaluated for single audio coding-decoding, using objective perceptual measures. The experimental results for MP3 coding exhibits an efficient restoration of the attacks and a significant improvement of the audio quality.

关键词： Transient analysis Detectors Digital audio players Decoding audio coding Discrete cosine transforms

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Multichannel audio coding and its applications in DAB and DVB

Multichannel audio coding and its applications in DAB and DV...

引用

International Conference on Signal Processing Proceedings (ICSP)

作者： M. Bosi Digital Theater Systems Inc. Agoura Hills CA USA

In spite of a steady increase in storage media capacity and transmission bandwidth, high quality, multichannel audio creates a challenge for traditional and new delivery media. This paper presents an overview of the technologies adopted in multichannel audio coding. New trends, including increased audio sample resolution and sampling rates, are discussed. Different channel configurations, their advantages and disadvantages, and their impact on system data rates and capacity are examined. Emerging formats like DAB and DVB are introduced. Finally, applications of multichannel audio in DAB and DVB are examined.

关键词： audio coding Digital video broadcasting Loudspeakers Frequency Bandwidth Sampling methods Board of Directors Marine technology Signal to noise ratio Dynamic range

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FLEXCODE - FLEXIBLE audio coding

FLEXCODE - FLEXIBLE AUDIO CODING

引用

IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Janusz Klejsa Minyue Li W. Bastiaan Kleijn ACCESS Linnaeus Center Electrical Engineering KTH - Royal Institute of Technology 10044 Stockholm Sweden

ISBN: (纸本)9781424442959;9781424442966

Modern networks are highly variable and, as a result, source coders are commonly used under conditions that they were not designed for. We address this problem with a source-coding philosophy that aims at the instantaneous re-optimization of a source coder to match a wide range of constraints on rate or quality and a wide range of packet-loss rates. We present a number of technologies that can be reconfigured by solving analytic relations that use the current conditions and a statistical description of the source as input. The technologies include distribution-preserving quantizers, flexible multiple-description quantizers, and a rate distribution scheme. Based on the generic technologies, we created a complete audio coder. Formal listening tests show that the resulting audio coding scheme with full flexibility provides a quality that is on-par with the best standardized codecs for any particular rate.

关键词： audio coding source modeling quantization audio coding Quantization audio coder audio FLEXIBLE Packet loss quantizer ENCODER wide-range

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Primary-Ambient Signal Decomposition and Vector-Based Localization for Spatial audio coding and Enhancement

Primary-Ambient Signal Decomposition and Vector-Based Locali...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Michael M. Goodwin Jean-Marc Jot Creative Advanced Technology Center Scotts Valley CA USA

Spatial audio coding and enhancement address the growing commercial need to store and distribute multichannel audio and to render content optimally on arbitrary reproduction systems. In this paper, we discuss a spatial analysis-synthesis scheme which applies principal component analysis to an STFT-domain representation of the original audio to separate it into primary and ambient components, which are then respectively analyzed for cues that describe the spatial percept of the audio scene on a per-tile basis; these cues are used by the synthesis to render the audio appropriately on the available playback system. The proposed framework can be tailored for robust spatial audio coding, or it can be applied directly to enhancement scenarios where there are no rate constraints on the intermediate spatial data and audio representation.

关键词： Signal resolution audio coding Principal component analysis Layout Loudspeakers Information analysis Signal synthesis Signal analysis Decoding Robustness

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Fine grain scalable perceptual and lossless audio coding based on IntMDCT

Fine grain scalable perceptual and lossless audio coding bas...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： R. Geiger A. Herre G. Schuller T. Sporer Fraunhofer AEMT Ilmenau Germany Fraunhofer IIS A Erlangen Germany

This papers presents an embedded fine grain scalable perceptual and lossless audio coding scheme. The enabling technology for this combined perceptual and lossless audio coding approach is the integer modified discrete cosine transform (IntMDCT), which is an integer approximation of the MDCT based on the lifting scheme. It maintains the perfect reconstruction property and therefore enables efficient lossless coding in the frequency domain. The close approximation of the MDCT also allows us to build a perceptual coding scheme based on the IntMDCT. In this paper a bitsliced arithmetic coding technique is applied to the IntMDCT values. Together with the encoded shape of the masking threshold a perceptually hierarchical bitstream is obtained, containing several stages of perceptual quality and extending to lossless operation when transmitted completely. A concept of encoding subslices is presented in order to obtain a fine adaptation to the masking threshold especially in the range of perceptually transparent quality.

关键词： audio coding Discrete cosine transforms MPEG 4 Standard Scalability Frequency domain analysis Arithmetic Masking threshold Codecs Noise shaping Shape

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HILN-the MPEG-4 parametric audio coding tools

HILN-the MPEG-4 parametric audio coding tools

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： H. Purnhagen N. Meine Lab. fur Informationstechnol. Hannover Univ. Germany Laboratorium für Informationstechnologie University of Hannover Hanover Germany

The MPEG-4 audio Standard combines tools for efficient and flexible coding of audio. For very low bitrate applications, tools based on a parametric signal representation are utilised. The parametric speech coding tools (HVXC) are already available in Version 1 of MPEG-4. The main focus of this paper is on the parametric audio coding tools "Harmonic and Individual Lines plus Noise" (KILN) which are included in Version 2 of MPEG-4. As already indicated by their name, the HILN tools are based on the decomposition of the audio signal into components which are described by appropriate source models and represented by model parameters. This paper gives an overview of the HILN tools, presents the recent advances in signal modelling and parameter coding, and concludes with an evaluation of the subjective audio quality.

关键词： MPEG 4 Standard audio coding Bit rate Speech coding Scalability Mobile communication Laboratories Signal representations Context Digital multimedia broadcasting

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A Memory-Based Hardware Accelerator for Real-Time MPEG-4 audio coding and Reverberation

A Memory-Based Hardware Accelerator for Real-Time MPEG-4 Aud...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Guo-An Jian Chih-Da Chien Jiun-In Guo Department of Computer Science and Information Engineering National Chung Hsing University Chiayi Taiwan

In this paper we propose a memory-based hardware accelerator for MPEG-4 audio coding and reverberation to achieve both high quality and reality of audio. The proposed design can realize both the computation-intensive component of 256/2048-point IMDCT and the 1024-point FFT-based reverberation through the same hardware engine by adopting a unified IMDCT/FFT/IFFT algorithm, which greatly reduces the hardware cost. The proposed design can achieve both the real-time 5.1 channel audio decoding at the sampling rate of 44.1 KHz and audio reverberation with the hardware cost of 26,633 gates and 4.6K words of local memory for storing transform coefficients and temporary results. The maximum working frequency achieves 220 MHz when implemented by UMC 0.18mum CMOS technology, which can fit the real-time processing requirement of many high quality MPEG-4 audio coding applications

关键词： Hardware MPEG 4 Standard audio coding Reverberation Costs CMOS technology Algorithm design and analysis Engines Decoding Sampling methods

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