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检索条件"主题词=Audio coding"

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Hybrid LPC and discrete wavelet transform audio coding with a novel bit allocation algorithm

Hybrid LPC and discrete wavelet transform audio coding with ...

引用

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

作者： S.D. Boland M. Deriche Signal Processing Research Centre School of Electrical and Electronic Systems Engineering Queensland University of Technology Brisbane QLD Australia

This paper examines a new method for coding high quality digital audio signals based on a combination of linear predictive coding (LPC) and the discrete wavelet transform (DWT). In this method, a linear predictor is first used to model each audio frame. Then, the prediction error is analyzed using the DWT. The LPC coefficients and DWT coefficients are quantized using a novel bit allocation scheme which minimizes the overall quantization error with respect to the masking threshold. The proposed coder is capable of delivering near-transparent audio signal quality at encoding bit rates of around 90-96 kb/s. Objective and subjective results suggest that the proposed coder operating at 90-96 kb/s has a performance comparable to that of the MPEG layer II codec operating at 128 kb/s.

关键词： Discrete wavelet transforms Linear predictive coding audio coding Bit rate Predictive models Error analysis Quantization Masking threshold Encoding Codecs

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AVS2 SPEECH AND audio coding SCHEME FOR HIGH QUALITY AT LOW BITRATES

AVS2 SPEECH AND AUDIO CODING SCHEME FOR HIGH QUALITY AT LOW ...

引用

IEEE International Conference on Multimedia and Expo Workshops

作者： Lin Jiang Ruimin Hu Xiaochen Wang Maoshen Zhang Zhongyuan, Wang National Engineering Research Center for Multimedia Software Computer College Wuhan University

ISBN: (纸本)9781479947164

With the increasing number of portable and wireless devices and the expanding of wireless network bandwidth, there is a growing demand for high quality and low bitrate speech and audio codec. This paper presents a speech and audio codec, which is based upon AVS P10. In our presented codec, ACELP and TCX technology is adopted, and also presents a hybrid bandwidth extension to coding high band signal. The paper gives an overview of the codec architecture and presents results of formal listening tests, objective test, and complicity of algorithm comparing this new codec with ITU-T G.718 and AVS P10. This result shows a comparable performance and forms the basis of the reference model in the ongoing AVS standardization activity for AVS2 speech and audio coding.

关键词： Speech coding audio coding AVS

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Lossless audio coding by predictor blending

Lossless audio coding by predictor blending

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International Conference on Telecommunications and Signal Processing (TSP)

作者： Grzegorz Ulacha Ryszard Stasiński Faculty of Computer Science & Information Systems West Pomeranian University of Technology Szczecin Poland Department of Electronics & Communications Poznan University of Technology Poznan Poland

In the paper potential bit rate savings obtained when implementing lossless audio coding using predictor blending technique are investigated. Three algorithms are analyzed: the simplest Blend-11, Blend-12, and Blend-13 containing 11, 12, and 13 subpredictors, respectively, and tested on 16 widely used benchmark recordings. Savings linked with inter-channel audio dependencies are also considered, appropriate predictor forms the part of Blend-13 method. Results for the last algorithms are also obtained when modeling stage is followed by an advanced arithmetical coder. It is shown that indeed, predictor blending approach to lossless audio compression improves data compaction properties of coding algorithms.

关键词： Prediction algorithms Bit rate Context audio coding Entropy Histograms

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Perceptual linear predictive noise modelling for sinusoid-plus-noise audio coding

Perceptual linear predictive noise modelling for sinusoid-pl...

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

作者： R.C. Hendriks R. Heusdens J. Jensen Department of Mediamatics Delft University of Technnology Delft Netherlands

Sinusoidal coding of an audio subject to a bit-rate constraint, in general, results in a noise-like residual signal. This residual signal is of high perceptual importance; reconstruction of audio using the sinusoidal representation only typically results in an artificial sounding reconstruction. We present a new method, called perceptual linear predictive coding (PLPC), where the residual is encoded by applying LPC in the perceptual domain. This method minimizes a perceptual modelling error and therefore represents only residual components that are of perceptual relevance, while automatically discarding components masked by the sinusoidally coded part. Subjective listening tests show that PLPC performs significantly better than ordinary LPC as a sinusoidal residual coding technique. Furthermore, PLPC combined with a flexible segmentation and model order allocation algorithm leads to a significant gain in terms of R/D performance for fragments with fast changing characteristics.

关键词： Predictive models audio coding Linear predictive coding Acoustic noise Encoding Testing Performance evaluation Performance gain Frequency Filter bank

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Aspects of current standardization activities for high-quality, low-rate multi-channel audio coding

Aspects of current standardization activities for high-quali...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： M. Bosi C. Todd T. Holman Dolby Laboratories Inc. San Francisco CA USA USC School of Cinema Television and Lucasfilm THX San Rafael CA USA

This paper analyzes directions in the current standardization activities for multi-channel audio, briefly reviews the composite coding schemes AC-3 and ISO 11172-3 compatible systems, and discusses requirements, features, and time-tables for the audio systems in the ISO/Moving pictures Expert Group (MPEG) phase 2 and the United States high definition television (HDTV) standardization processes.< >

关键词： Standardization audio coding Motion pictures TV HDTV Loudspeakers Optical films Laboratories audio systems Frequency

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MPEG-4 Scalable to Lossless audio coding - Emerging International Standard for Digital audio Compression

MPEG-4 Scalable to Lossless Audio Coding - Emerging Internat...

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IEEE Workshop on Multimedia Signal Processing

作者： R. Yu X. Lin S. Rahardja H. Huang Institute for Infocomm Research Singapore

Recently, with the advance of network and storage technologies, it is becoming realistic that people will enjoy high sampling rate, high resolution audio contents with lossless quality. Envisioning of such a need, the international standardization body MPEG has recently introduced a scalable tool for lossless audio coding, namely, MPEG-4 audio Scalable to Lossless (SLS) coding. MPEG-4 SLS integrates the functionalities of lossless audio coding, perceptual audio coding, and fine granular scalable audio coding in a single framework; meanwhile it provides backward compatibility to MPEG-4 Advanced audio coding (AAC) at the bit-stream level. This new tool, in combination with the existing MPEG audio toolset, provides a universal digital audio format that can be used in a variety of application domains such as professional audio, Internet music, consumer electronics, broadcasting

关键词： MPEG 4 Standard audio coding audio compression Laser sintering Sampling methods Standardization Transform coding Internet Consumer electronics Digital audio broadcasting

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Complexity scalable audio coding algorithm based on wavelet packet decomposition

Complexity scalable audio coding algorithm based on wavelet ...

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International Conference on Signal Processing Proceedings (ICSP)

作者： He Dongmei Gao Wen Wu Jiangqin Dept. of Comput. Sci. & Eng. Harbin Inst. of Technol. China Department of Computer Science and Engineering Harbin Institute of Technology Harbin

In this paper, a complexity scalable audio coding algorithm based on wavelet packet decomposition is proposed. The incomplete wavelet packet decomposition with the property of complexity scalability to the signals is described in this paper. The zerotree coding of the coefficients is implemented which uses the characteristics of human ears and the self-similarity among different subbands wavelet coefficients. This algorithm cannot only be used to achieve transparent quality reconstruction audio at low bit rate, but also enable to support complexity scalable coding and multirate scalable coding. audio encoding and decoding can be realized in real-time utilizing this algorithm on computers with different levels of computational power. The experimental results indicate that this algorithm is of practical value to the real-time multimedia applications.

关键词： audio coding Wavelet packets Scalability Humans Ear Wavelet coefficients Bit rate Encoding Decoding Application software

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Sinusoidal modeling of spot microphone signals based on noise transplantation for multichannel audio coding

Sinusoidal modeling of spot microphone signals based on nois...

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European Signal Processing Conference (EUSIPCO)

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

This paper focuses on high-fidelity multichannel audio modeling 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 model 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 (microphone signals before the mixing stage) can be obtained from the noise part of one of the signals (reference, which is fully encoded), by transforming the reference noise part using the noise envelope of each of the remaining signals.

关键词： Noise Microphones Computational modeling Bit rate audio coding Harmonic analysis

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Lossless audio coding with MPEG-4 structured audio

Lossless audio coding with MPEG-4 structured audio

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International Conference on Web Delivering of Music (WEDELMUSIC)

作者： N. Vasiloglou R.W. Schafer M.C. Hans Center for Signal and Image Processing Georgia Institute of Technology Atlanta GA USA Mobile & Media Systems Laboratory Hewlett Packard Laboratories Palo Alto CA USA

MPEG-4 structured audio (SA) has been proposed as a flexible standard for generalized audio coding. Originating out of Netsound software developed at MIT SA is based on MIDI-synthesis of sound, but it is enriched with DSP algorithms so as to allow emulation of other types of coders designed for speech and audio signals. We have investigated the use of structured audio for lossless coding of audio signals, and have found that certain limitations of structured audio make implementations of lossless coders less straightforward than might be desired. In particular we have used SA to implement an MPEG-4 compliant version of the lossless audio coder audioPaK. To implement and validate our new coder we used the software system Sfront, which translates MPEG-4 SA files into efficient C programs that render the audio signal.

关键词： audio coding MPEG 4 Standard Digital signal processing Software algorithms Signal processing Image processing Laboratories Emulation Signal design Algorithm design and analysis

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Multichannel audio coding and its applications in DAB and DVB

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

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International Conference on Communication Technology (ICCT)

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 (digital audio broadcasting) and DVB (digital video broadcasting) are introduced. Finally, applications of multichannel audio in DAB and DVB are examined.

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

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