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检索条件"主题词=Audio coding"

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2003 IEEE Workshop on Applications of Signal Processing to audio and Acoustics (IEEE Cat. No.03TH8684)

2003 IEEE Workshop on Applications of Signal Processing to A...

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

The following topics are dealt with: analysis and synthesis - filter and filterbank design, sinusoidal and noise modeling; source separation and system identification; arrays and beamformers; active noise cancellation; music signal processing - music transcription, beat and tempo tracking, segmentation, classification, and music information retrieval; audio coding; physical modeling; speech, singing, and vocal tract modeling; virtual acoustics - binaural audio and HRTF processing; music resynthesis and cross-synthesis.

关键词： audio coding Acoustic signal processing Filtering Identification Array signal processing Interference suppression Music Information retrieval Biological system modeling

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A unified fast computation of the evenly and oddly stacked MDCT/MDST

A unified fast computation of the evenly and oddly stacked M...

引用

IEEE Region 8 International Symposium on Video/Image Processing and Multimedia Communications (EURASIP)

作者： V. Britanak Institute or Informatics Slovak Academy of Sciences Bratislava Slovakia

A unified approach to the fast computation of evenly and oddly stacked modified discrete cosine transform (MDCT) and modified discrete sine transform (MDST) is presented. Although the evenly and oddly stacked MDCT/MDST are quite different filter banks, there exists an intimate relation between them, and consequently, the efficient oddly stacked MDCT/MDST computation can be realized via the evenly stacked MDCT/MDST and vice versa only by simple preprocessing and postprocessing of input and output data sequences. In particular, it is shown that the transposed evenly and oddly stacked MDCT and MDST matrices are actually the generalized inverses or pseudoinverses of their corresponding forward transform matrices.

关键词： Filter bank Discrete cosine transforms Discrete transforms audio coding audio compression Informatics Image reconstruction Videoconference Video compression Image processing

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A progressive to lossless embedded audio coder (PLEAC) with reversible modulated lapped transform

A progressive to lossless embedded audio coder (PLEAC) with ...

引用

IEEE International Conference on Multimedia and Expo (ICME)

作者： Jin Li Redmond WA USA

ISBN: (纸本)0780376633

A progressive to lossless embedded audio coder (PLEAC) has been proposed. PLEAC is based purely on reversible transform, which is designed to mimic the non-reversible transform in a normal psychoacoustic audio coder as much as possible. Coupled with a high performance embedded entropy codec, this empowers PLEAC with both lossless capability and fine granular scalability. The PLEAC encoder generates a bitstream that if fully decoded, completely recovers the original audio waveform without loss. Moreover, it is possible to scale this bitstream in a very large bitrate range, with granularity down to a single byte. Extensive experimental results support the superior lossless performance and bitstream scalability of the PLEAC coder.

关键词： Modulation coding Scalability Performance loss Entropy Codecs Decoding audio coding Psychology audio compression Filters

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Improved low bit-rate audio compression using reduced rank ICA instead of psychoacoustic modeling

Improved low bit-rate audio compression using reduced rank I...

引用

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

作者： A. Ben-Shalom M. Werman S. Dubnov School of Computer Science Hebrew University Jerusalem Israel Ben-Gurion University of the Negev Beersheba Israel

Traditional audio coding is based on a perceptual compression paradigm that exploits psychoacoustic information to efficiently encode audio signals. Recently, extensive research has been conducted in order to understand how the brain encodes natural signals. These results suggest that the encoding process is very efficient in terms of redundancy reduction of the signal information. It could be that the psychoacoustic effects (such as the masking effect) are only a special case of a more general redundancy reduction mechanism that exists in the auditory pathway. Motivated by this work we propose a new audio coding scheme that is based on improved sound representation found by independent component analysis. Using a local linear, low rank, nonorthogonal transform, we remove additional redundancies in the signal. At low bitrates this coding scheme gives results superior to a legacy perceptual encoding scheme for different kinds of audio signals.

关键词： audio compression Independent component analysis Psychology Psychoacoustic models Humans Auditory system Bit rate audio coding Redundancy Ear

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A multiresolution ABS/OLA sinusoidal model using wavelet transform

A multiresolution ABS/OLA sinusoidal model using wavelet tra...

引用

International Symposium on Signal Processing and Its Applications (ISSPA)

作者： Ki-Hong Kim In-Ho Hwang National Security Research Institute Daejeon South Korea

ISBN: (纸本)0780379462

The sinusoidal model has been applied to a broad range of speech and audio coding, such as analysis/synthesis, time-and frequency-scale modification, fundamental frequency modification, speech enhancement, and co-channel separation. In this paper, we propose a multiresolution analysis-by-synthesis/overlap-add (ABS/OLA) sinusoidal model using a wavelet transform. In the proposed scheme, after decomposing an input speech signal into multiresolution subband signals using the wavelet transform, classical ABS/OLA sinusoidal models with different window lengths are applied to each subband signals respectively. It is shown that by applying a proper-sized analysis window, much more accurate sinusoidal components can be estimated. Experimental results have shown that the proposed multiresolution ABS/OLA sinusoidal model can achieve better performance than that of the classical ABS/OLA sinusoidal model in terms of the spectral characteristics, phase characteristics, and the quality of synthetic speech.

关键词： Wavelet transforms Signal resolution Speech enhancement Speech synthesis Speech analysis Frequency synthesizers Speech coding audio coding Signal synthesis Wavelet analysis

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A low power VLSI implementation for variable length decoder in MPEG-1 layer III

A low power VLSI implementation for variable length decoder ...

引用

IEEE International Conference on Multimedia and Expo (ICME)

作者： Tsung-Han Tsai Wen-Cheng Chen Chun-Nan Liu Department of Electrical Engineering National Central University Chungli Taiwan

ISBN: (纸本)0780379659

MPEG layer III (MP3) audio coding algorithm was a widely used audio coding standard. It involves several complex coding techniques and is therefore difficult to create an efficient architecture design. The variable length decoding was an important part, which needs great amount of search and memory read/write operations. In this paper a data driven variable length decoding algorithm is presented, which can exploit the signal statistics of variable length codes to reduce power. The decoder was designed based on simplicity and low-cost, low power consumption while retaining the high efficiency requirements.

关键词： Very large scale integration Decoding Digital audio players Table lookup audio coding Binary trees Detectors Energy consumption Transform coding audio compression

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Integer fast modified cosine transform

Integer fast modified cosine transform

引用

IEEE International Conference on Multimedia and Expo (ICME)

作者： Dong-Yan Huang Ruihua Ma Signal Processing Program/Video Processing Group Institute for Infocomm Research Singapore

ISBN: (纸本)0780379659

In this paper, an efficient implementation of the forward and inverse MDCT is proposed for even-length MDCT. The algorithm uses discrete cosine transform of type II (DCT-II) to compute the forward MDCT and their inverse DCT-III to compute the inverse MDCT. The lifting scheme is used to approximate multiplications appearing in the MDCT lattice structure where the dynamic range of the lifting coefficients can be controlled by proper choices of the lifting factorizations. The new structure requires less multiplications and additions than previous reported algorithms. The new transform has the properties that it is an integer-to-integer mapping and is reversible. Moreover, it inherits most of the attractive properties of the MDCT, including a good spectral representation of the audio signal, critical sampling and overlapping of blocks. Hence, the intMDCT is suitable for both lossless and lossy audio coding.

关键词： Discrete cosine transforms audio coding Sampling methods Lattices Discrete transforms Signal processing Dynamic range Signal processing algorithms Predictive models Redundancy

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Robust audio watermarking using a chirp based technique

Robust audio watermarking using a chirp based technique

引用

IEEE International Conference on Multimedia and Expo (ICME)

作者： S. Erkucuk S. Krishnan M. Zeytinoglu Department of Electrical and Computer Engineering Ryerson University Toronto ONT Canada

In this study, we propose a new spread spectrum audio watermarking algorithm that embeds linear chirps as watermark messages. Different chirp rates, i.e., slopes on the time-frequency (TF) plane, represent watermark messages such that each slope corresponds to a different message. We extract the watermark message using a line detection algorithm based on the Hough-Radon transform (HRT). The HRT detects the directional elements that satisfy a parametric constraint in the image of a TF plane. The proposed method not only detects the presence of watermark, but also extracts the embedded watermark bits and ensures the message is received correctly. The results show that the HRT detects the embedded watermark message even after common signal processing operations such as MPEG audio coding, resampling, lowpass filtering and amplitude re-scaling.

关键词： Robustness Watermarking Chirp Spread spectrum communication Signal processing algorithms Time frequency analysis Detection algorithms Signal processing audio coding Filtering

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Phase-based note onset detection for music signals

Phase-based note onset detection for music signals

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： J.P. Bello M. Sandler University of London London UK

Summary form only given. Note onsets mark the beginning of attack transients, short areas of a note containing rapid changes of the signal spectral content. Detecting onsets is not trivial, especially when analysing complex mixtures. Applications for note onset detection systems include time stretching, audio coding and synthesis. An alternative to standard energy-based onset detection is proposed by using phase information. It is suggested that by observing the frame-by-frame distribution of differential angles, the precise moment when onsets occur can be detected with accuracy. Statistical measures are used to build the detection function. The system is tested and tuned on a database of complex recordings.

关键词： Multiple signal classification Phase detection Performance analysis Signal detection audio coding Signal synthesis System testing Databases

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Perceptually transparent audio compression based on a variable bit rate AAC coder

Perceptually transparent audio compression based on a variab...

引用

IEEE Region 8 International Symposium on Video/Image Processing and Multimedia Communications (EURASIP)

作者： A. Szwabe C. Jedrzejek Institute of Electronics and Telecommunications Poznan University of Technology Poznan Poland University of Agriculture and Technology Bydgoszcz Poland

The paper presents an implementation of a perceptually transparent variable bit rate (VBR) audio coder that complies with the MPEG-4 advanced audio coding (AAC) standard. The coder is based on the MPEG-4 reference software source code of CBR coder. With small modifications in the reference source code a coder guaranteeing perceptual transparency is achieved at compression ratios considerably better than that of the reference CBR coder. Comparison tests indicate that files generated by the perceptually transparent VBR coder applied to a standard set of test items are considerably smaller (in average 24%) than the smallest CBR files of the same quality.

关键词： audio compression Bit rate Image coding MPEG 4 Standard Code standards audio coding Testing Reservoirs Information technology Agriculture

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