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检索条件"主题词=Audio coding"

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Non-stationary signal parametric modeling techniques with an application to low bit rate audio coding

Non-stationary signal parametric modeling techniques with an...

引用

International Conference on Signal Processing Proceedings (ICSP)

作者： R. Boyer S. Essid N. Moreau Department of Signal and Image Processing ENST Paris France

Low bit rate audio coding often relies on Fourier representation despite its limitations for transient signal modeling. This study proposes alternative decompositions and expansion strategies that lead to more accurate modeling. Two classes of methods are considered, subspace decomposition methods, and atomic decomposition methods and their performances are compiled to propose an audio modeling scheme amenable to low bit rate coding.

关键词： Parametric statistics Bit rate audio coding Matching pursuit algorithms Signal processing Image coding Transient analysis Signal analysis Fading Acoustic noise

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AMR-WB+: a new audio coding standard for 3rd generation mobile audio services

AMR-WB+: a new audio coding standard for 3rd generation mobi...

引用

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

作者： J. Makinen B. Bessette S. Bruhn P. Ojala R. Salami A. Taleb Multimedia Technologies Laboratory Nokia Research Center Finland University of Sherbrooke QUE Canada Multimodal Technologies Inc. Sweden Voice Age Corporation Montreal QUE Canada

Highly efficient low-rate audio coding methods are required for new compelling and commercially interesting applications of streaming, messaging and broadcasting services using audio media in 3rd generation mobile communication systems. After an audio codec selection phase, 3GPP has standardized the extended AMR-WB (AMR-WB+) codec that provides a unique performance at very low bit rates from below 10 kbps up to 24 kbps. This paper discusses the requirements imposed by mobile audio services and gives a technology overview of AMR-WB+ as a codec matching these requirements while providing outstanding audio quality.

关键词： audio coding Codecs Streaming media Bit rate Testing Multimedia communication Digital multimedia broadcasting Multimedia systems Laboratories 3G mobile communication

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Combined speech and audio coding by discrimination

Combined speech and audio coding by discrimination

引用

IEEE Workshop on Speech coding Proceedings

作者： L. Tancerel S. Ragot V.T. Ruoppila R. Lefebvre Department of Electrical Engineering University of Sherbrooke Sherbrooke QUE Canada

We propose in this paper a general solution for combined speech and audio coding. Particularly, we describe a speech/music discrimination procedure for multi-mode wideband coding. The speech/music decision is updated only when a low-energy frame is detected, and kept unchanged otherwise. The signal is classified using second-order statistics of discriminant parameters. An experimental CELP/transform coder operating at 16 kbit/s is demonstrated. Results show improved performance when compared to single-mode encoding.

关键词： Speech coding audio coding Multiple signal classification Wideband Switches Delay estimation Statistics Bit rate Forward contracts Linear predictive coding

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Transform audio coding with Arithmetic-Coded Scalar Quantization and Model-Based Bit Allocation

Transform Audio Coding with Arithmetic-Coded Scalar Quantiza...

引用

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

作者： Marie Oger Stephane Ragot Marc Antonini France Télécom Research and Development TECH/SSTP/RVA Lannion France Laboratory 13S-UMR 6070 CNRS CNRS and University of Nice-Sophia Antipolis Sophia-Antipolis France

In this paper we present a new model-based method to code the transform coefficients of audio signals. The histogram of transform coefficients is approximated by a generalized Gaussian model for efficient model-based bit allocation and the spectrum is coded by scalar quantization followed by arithmetic coding. An example coder operating at 16 kHz and using predictive modified discrete cosine transform (MDCT) coding is described. We compare the performance of the proposed coder with ITU-T G.722.1. Objective and subjective quality results are presented. The proposed coder is better than ITU-T G.722.1 at 24 kbit/s and equivalent at 32 kbit/s.

关键词： audio coding Quantization Bit rate Shape Speech coding Discrete transforms Histograms Arithmetic Discrete cosine transforms Random variables

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Spectral Band Replication Tool for Very Low Delay audio coding Applications

Spectral Band Replication Tool for Very Low Delay Audio Codi...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Tobias Friedrich Gerald Schuller Fraunhofer Institute for Digital Media Technology Ilmenau Germany

In this paper a Spectral Band Replication (SBR) tool for low delay audio applications is presented. One goal of this enhancement tool is to reduce the needed bit rate for the representation of audio data using an arbitrary audio codec. Another goal is to keep the algorithmic delay as low as possible. A low coding delay is essential for instance for real time applications like distributed music production under live conditions or telephone conferencing. The low delay SBR approach proposed in this paper uses techniques developed for speech coding purposes and is associated with artificial bandwidth extension methods, particularly spectral folding. Further, the tool exclusively operates in the time domain using prediction methods and adaptive filters in order to avoid additional delay which can be caused by using a filter bank.

关键词： audio coding Delay effects Bit rate Codecs Telephony Speech coding Bandwidth Prediction methods Adaptive filters Added delay

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A high quality re-quantization/quantization method for MP3 and MPEG-4 AAC audio coding

A high quality re-quantization/quantization method for MP3 a...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Tsung-Han Tsai Chuh-Chu Yen Department of Electrical Engineering National Central University Taiwan

ISBN: (纸本)0780374487

MP3 is the popular audio coding standard. But now, a new higher quality audio coding standard Advanced audio coding (AAC) is proposed and widely used. The quantization/re-quantization is essential in both MP3 and AAC. This paper proposes a new high accuracy estimation algorithm for MP3 and MEPG-4 AAC audio coding. The algorithm can be applied not only for re-quantization process in decoder, but also for quantization in encoder. According to the floating-point behavior description, the proposed algorithm can achieve the lowest error result measured in quantization/re-quantization. Furthermore, in order to maintain the high quality with limited data precision, finite word length for fixed-point effects are investigated. Our algorithm has the low complexity advantages of multiplication-free and low memory requirements which will be more suitable in VLSI design for audio encoder/decoder.

关键词： Quantization Digital audio players MPEG 4 Standard audio coding Table lookup Decoding Costs Hardware Very large scale integration Algorithm design and analysis

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New results in low bitrate audio coding using a combined harmonic-wavelet representation

New results in low bitrate audio coding using a combined har...

引用

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

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

In this paper, we propose a new combined harmonic-wavelet representation for audio where a harmonic analysis-synthesis scheme is used, first, to approximate each audio frame as a sum of several sinusoids. Then, the difference between the original signal and the reconstructed harmonic signal is analyzed using a wavelet filtering scheme. After each step (harmonic analysis and wavelet filtering), parameters are quantized and encoded. Compared to previously proposed methods, our audio coder uses different harmonic analysis-synthesis and wavelet filtering schemes. We use the total least squares (TLS)-prony algorithm for the harmonic analysis-scheme, and an M-band wavelet transform for analyzing the residual. Altogether, our proposed coder is capable of delivering excellent audio signal quality at encoder bitrates of 60-70 kb/s.

关键词： Bit rate audio coding Harmonic analysis Power harmonic filters Wavelet analysis Filtering Algorithm design and analysis Least squares approximation Signal analysis Least squares methods

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High quality audio coding using a novel hybrid WLP-subband coding algorithm

High quality audio coding using a novel hybrid WLP-subband c...

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Asia-Pacific Conference on Communications

作者： Yu Rongshan Ko Chi Chung Digital System and Application Lab Department of Electrical Engineering National University of Singapore Singapore

ISBN: (纸本)1864354518

A high quality audio coding scheme based on a novel hybrid algorithm combining warped linear prediction (WLP) and subband coding is proposed. The proposed codec is capable of providing a high quality audio output at a low bit-rate. Subjective tests have shown that the proposed codec is able to provide a performance comparable to that of the MPEG audio layer II while operating at a lower bit-rate.

关键词： audio coding Frequency Masking threshold Quantization Linear predictive coding Filters Low-frequency noise Codecs Humans Auditory system

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Performance Analysis and Evaluation of AVS-M audio coding

Performance Analysis and Evaluation of AVS-M Audio Coding

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2010 International Conference on audio,Language and Image Processing(2010年音频、语言与图像处理国际会议 ICALIP 2010)

作者： Zhang Wen Zhang Tao Zhao Liang Zhang Hai School of Electronic and Information Engineering Tianjin University China

ISBN: (纸本)9781424458653

AVS-M audio standard, which targets at wireless network and mobile equipment, is now independently drawn up in China. Its framework is similar to that of AMR-WB+. The performance of AVS-M audio core algorithms is analyzed in this paper. In order to analyze its complexity fixed-point version of AVS-M codec are implemented on DSP platform. At last, a performance evaluation between AVS-M and AMR-WB+ is discussed. The result is that AVS-M audio performance is no worse than AMR-WB+ on average.

关键词： AMR-WB Mobile devices Performance Evaluation audio coding audio Wireless networks

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Perceptually Prioritized Bit-Plane coding for High-Definition Advanced audio coding

Perceptually Prioritized Bit-Plane Coding for High-Definitio...

引用

IEEE International Symposium on Multimedia

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

ISBN: (纸本)0769527469;9780769527468

Wide bitrate range scalability is now the latest trend in audio coding. A lot of efforts has been devoted to the development of algorithms for more efficient scalable audio coder that scales from very low bitrate. Scalable audio coding technique such as MPEG-4 scalable lossless coding (SLS) offers a unified solution for high-compression perceptual audio and high-quality lossless audio. SLS provides a fine-grain scalable extension of the well-known MPEG-4 advanced audio coding (AAC) perceptual audio coder up to fully lossless reconstruction. Recently, the combination of SLS and AAC coder is renamed as "high definition advanced audio coding" (HD-AAC). It is observed that HD-AAC can be further improved at intermediate enhancement bitrate when the core bitrate is low. In this paper, a perceptually prioritized bit-plane coding (PPBPC) is proposed. With this novel coding scheme, the bit-plane coding is performed with priorities according to the perceptual information of the signal to be coded. By using this low-complexity structure with trivial extra side information, the bit-plane coding for scalable audio can be implemented in a perceptually more efficient manner and the quality of the audio under aforementioned scenario is greatly improved

关键词： audio coding Bit rate Laser sintering MPEG 4 Standard Signal resolution Internet Encoding Tellurium Scalability Sampling methods

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