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检索条件"主题词=Audio coding"

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A perception-based audio watermarking scheme

A perception-based audio watermarking scheme

引用

IEEE Region 10 International Conference TENCON

作者： B. Lei I. Y. Soon School of Electrical and Electronic Engineering Nanyang Technological University Singapore

In this paper, a perceptual audio watermarking scheme for MPEG scalable lossless audio that operates directly in the compressed data is proposed. The watermarking scheme is implemented in the compressed stream by modifying the quantized coefficients to satisfy the requirements of robustness, security and imperceptibility. The watermarking procedure directly exploits perceptual frequency masking of the human auditory system (HAS) to guarantee that the scheme is inaudible and robust. The watermark is constructed by selecting and quantizing the perceptually significant audio segments and coefficients. Extensive experimental results show that the high robustness and transparency of watermarked audio can be achieved simultaneously.

关键词： Watermarking Chaos Robustness Psychoacoustic models audio coding Discrete Fourier transforms Discrete wavelet transforms Transform coding Data security Discrete cosine transforms

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Rate distribution between model and signal for multiple descriptions

Rate distribution between model and signal for multiple desc...

引用

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

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

We consider the rate allocation problem for multiple-description quantization of the signal described by an adaptive model with a fixed structure. The source modeling in coding generally results in a two-stage description of the data, where one of the stages describes the model parameters, and the other describes the signal. Such a setup implies the existence of a trade-off between the rate spent on the parameters and the rate spent on the signal. We optimize this trade-off analytically for the multiple-description case using a method inspired by Minimum Description Length principle. We also provide an algorithm for optimizing the rate allocation between the components of the model-based multiple description coder. Finally we experimentally confirm our results. Our method facilitates the rate-adaptive multiple-description coding.

关键词： Vector quantization Lattices Encoding Optimization methods Context modeling Statistics audio coding Distortion Shape Entropy

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A FULL-BANDWIDTH audio CODEC WITH LOW COMPLEXITY AND VERY LOW DELAY

A FULL-BANDWIDTH AUDIO CODEC WITH LOW COMPLEXITY AND VERY LO...

引用

European signal processing conference

作者： Jean-Marc Valin Timothy B. Terriberry Gregory Maxwell Octasic Semiconductor 4101 Molson Street suite 300 Montreal (Quebec) Canada*** Foundation *** Foundation

ISBN: (纸本)9781617388767

We propose an audio codec that addresses the low-delay requirements of some applications such as network music performance. The codec is based on the modified discrete cosine transform (MDCT) with very short frames and uses gain-shape quantization to preserve the spectral envelope. The short frame sizes required for low delay typically hinder the performance of transform codecs. However, at 96 kbit/s and with only 4 ms algorithmic delay, the proposed codec out-performs the ULD codec operating at the same rate. The total complexity of the codec is small, at only 17 WMOPS for real-time operation at 48 kHz.

关键词： codecs audio coding REAL TIME OPERATION mdct(modified discrete cosine transform) frame size audio delay

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VLSI design of resource shared complex-QMF bank for HE-AAC decoder

VLSI design of resource shared complex-QMF bank for HE-AAC d...

引用

International Conference on ASIC

作者： Junqiao Huang Gaoming Du Duoli Zhang Yukun Song Luofeng Geng Minglun Gao VLSI Research Institute Hefei University of Technology Hefei China

A VLSI design of complex Quadrature Mirror Filterbank (QMF) for MPEG-4 High Efficiency Advanced audio coding (MPEG-4 HE-AAC) decoder using resource-sharing technique is proposed. The algorithm that uses conventional discrete cosine transform of type IV(DCT-IV) to optimize complex-QMF is derived in this paper. By using the proposed algorithm, the VLSI design of complex valued analysis quadrature mirror filterbank (complex-AQMF) and synthesis quadrature mirror filterbank (complex-SQMF) can improve resource efficiently by sharing the same DCT module. Experiment results show that the computational complexity of the complex-QMF can be reduced up to 8.59%, the VLSI architecture of the proposed algorithm can save about 53% of area and 50% memory due to the shared resources of DCT-IV.

关键词： Very large scale integration Decoding Mirrors Filter bank MPEG 4 Standard Discrete cosine transforms Algorithm design and analysis audio coding Computational complexity Computer architecture

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An improved psycho-acoustic model for MPEG 1 using a Morlet Cambridge wavelet

An improved psycho-acoustic model for MPEG 1 using a Morlet ...

引用

International Conference on Signals, Circuits and Systems, SCS

作者： Khalil Abid Kais Ouni Laboratory of Systems and Signal Processing LSTS National Engineering School of Tunis Tunis Tunisia

ISBN: (纸本)9781424443970;9781424443987

Many existing MP3 encoder employ an FFT-based transform to derive a spectral decomposition of the audio signal into uniform subbands with equal bandwidths. The nonuniform spectral resolution of the auditory system is taken into account. The initial audio signal processing within the psycho-acoustic model consists of a spectral decomposition to account for the frequency selectivity of the auditory system. However, the auditory system performs a nonuniform spectral decomposition of the acoustic signal in the cochlea. This paper presents a psycho-acoustic model based on an efficient nonuniform cochlear filter bank following the Cambridge model. Results of the proposed psycho-acoustic model applied to audio coding show better performance in terms of compression ratio and sound quality in comparison with the classical FFT model.

关键词： Psychology Psychoacoustic models Auditory system Digital audio players Bandwidth Signal resolution Acoustic signal processing Frequency Filter bank audio coding

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Strategies for bit allocation reuse in audio transcoding

Strategies for bit allocation reuse in audio transcoding

引用

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

作者： Mohamed F. Mansour Texas Instruments Inc. Dallas TX USA

ISBN: (纸本)9781424423538

We study the reuse of the bit allocation information in audio transcoding by exploiting the similarity in subband audio coding schemes. We show that important information can be deduced to reduce the encoder complexity even if the two coders employ different psychoacoustic model. We give a case study on MPEG AAC/Dolby AC-3 transcoding. The proposed algorithms can be extended to other audio transcoding schemes.

关键词： Bit rate Transcoding Quantization Psychoacoustic models Frequency estimation audio coding Additive noise Random variables Decoding Instruments

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AAC level controller on bitstream domain

AAC level controller on bitstream domain

引用

International Symposium on Communications and Information Technologies (ISCIT)

作者： Unggul Joo Hochong Park Department of Electronics Engineering Kwangwoon University Seoul South Korea

In digital broadcasting services, a difference in audio level among channels and contents causes users to modify audio output level whenever they change the channels or contents. To solve this problem, a novel method to control audio level of MPEG-2/4 AAC on the bitstream domain is proposed. It is verified that the proposed method has better performance than the level control in waveform domain.

关键词： Quantization Decoding Level control Digital audio broadcasting Bit rate Equations audio coding

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Speaker Identification and Verification from audio Coded Speech in Matched and Mismatched Conditions

Speaker Identification and Verification from Audio Coded Spe...

引用

2009 IEEE International Conference on Robotics and Biomimetics(2009 IEEE 机器人与仿生技术国际会议 ROBIO 2009)

作者： Tao Jiang Boyang Gao Jiqing Han Speech Research School of Computer Science and TechnologyHarbin Institute of TechnologyHarbin

ISBN: (纸本)9781424447749

We investigate the effect of audio coding on speaker identification and verification when training and testing conditions are matched and mismatched. Experiments use popular audio coding algorithms (Windows Media audio 9.1, Advanced audio coding, MPEG audio Layer III) and a speaker identification and verification system based on Gaussian mixture models. There is some loss in identification and verification performance for audio coding process without the change of sample rate, and a great loss when sample rate changes during audio coding process.

关键词： speaker recognition audio coding audio sample rate Verification Speech verification system MISMATCH

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Applications of signal analysis using autoregressive models for amplitude modulation

Applications of signal analysis using autoregressive models ...

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Sriram Ganapathy Samuel Thomas Petr Motlicek Hynek Hermansky Department of Electrical and Computer Engineering Johns Hopkins University USA Idiap Research Institute Martigny Switzerland

Frequency domain linear prediction (FDLP) represents an efficient technique for representing the long-term amplitude modulations (AM) of speech/audio signals using autoregressive models. For the proposed analysis technique, relatively long temporal segments (1000 ms) of the input signal are decomposed into a set of sub-bands. FDLP is applied on each sub-band to model the temporal envelopes. The residual of the linear prediction represents the frequency modulations (FM) in the sub-band signal. In this paper, we present several applications of the proposed AM-FM decomposition technique for a variety of tasks like wide-band audio coding, speech recognition in reverberant environments and robust feature extraction for phoneme recognition.

关键词： Signal analysis Amplitude modulation Frequency modulation Speech recognition Frequency domain analysis Predictive models Wideband audio coding Robustness Feature extraction

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Design of Block Lifting-Based Discrete Cosine Transform Type-II and IV

Design of Block Lifting-Based Discrete Cosine Transform Type...

引用

IEEE Digital Signal Processing Workshop

作者： Taizo Suzuki Masaaki Ikehara Department of Electronics and Electrical Engineering Keio University Yokohama Kanagawa Japan

ISBN: (纸本)9781424436767

This paper presents a novel lifting factorization of discrete cosine transform type-II and IV (DCT-II and IV). Although some conventional integer DCT-IIs (IntDCT-IIs) with block size 8 have been proposed, they are not generalized as arbitrary block size M. Using block lifting factorization which has an efficient structure for lossless-to-lossy image coding, we present IntDCT-IIs and IVs with arbitrary block size M that is called block lifting-based DCT-IIs and IVs (BLDCT-IIs and IVs). Finally, the validity of our method is proved by showing the results of lossless-to-lossy image coding in the most general case of the block size 8 and the extended size 16.

关键词： Discrete cosine transforms Image coding Transform coding audio coding Signal analysis Signal synthesis Video compression Discrete transforms Video coding

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