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检索条件"主题词=Audio coding"

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Towards optimal quantization in multistage audio coding

Towards optimal quantization in multistage audio coding

引用

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

作者： R. Vafin W.B. Kleijn Speech Processing Group Department of Signals Sensors and Systems KTH Royal Institute of Technology Stockholm Sweden

We develop a new method for quantization in multistage audio coding. We consider the case of a two-stage sinusoidal/waveform coder. Given a distortion measure and a bit-rate constraint, we analytically derive the optimal rate distribution between subcoders (stages) and the corresponding optimal quantizers, which allows the coder to adapt easily to changes in bit-rate requirements. We verify that the performance, both in terms of signal-to-noise ratio (SNR) and perceptual quality, is higher if the input to the second stage is obtained by subtracting the quantized first-stage reconstruction from the original signal, as opposed to subtracting the unquantized reconstruction.

关键词： Quantization audio coding Distortion measurement Bit rate Speech enhancement Speech processing Signal processing Sensor systems Signal to noise ratio Speech coding

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Approximation error analysis for transform-based lossless audio coding

Approximation error analysis for transform-based lossless au...

引用

IEEE Conference and Exhibition on Global Telecommunications (GLOBECOM)

作者： Y. Yokotani S. Oraintara R. Geiger G. Schuller K.R. Rao Electrical Engineering Department University of Technology Arlington TX USA Fraunhofer Institute for Digital Media Technology Ilmenau Germany

The integer modified discrete cosine transform (IntMDCT), an integer approximation of the MDCT, is a reversible transform realized by the lifting scheme and thus is a useful transform for lossless audio coding. Because of the integer approximation, however, the approximation error appears as a "noise floor" in the transform domain and limits the lossless coding efficiency. In this paper, a theoretical analysis of the approximation error of the IntMDCT is discussed. The result is then used to design a simple test filter applied to each rounding operation of the IntMDCT in such a way that the error spectrum is shaped towards the low frequencies. As a result, especially when the spectral energy of an input signal is concentrated in the low frequency domain, the lossless coding efficiency is improved.

关键词： Approximation error Error analysis audio coding Noise shaping Frequency Discrete transforms Testing Filters Noise cancellation Acoustic noise

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

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

引用

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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Extended linear prediction tools for lossless audio coding

Extended linear prediction tools for lossless audio coding

引用

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

作者： T. Moriya D.T. Yang T. Liebchen NTT Cyber Space Laboratories Tokyo Japan University of Southern California Los Angeles USA Technical University Berlin Berlin Germany

Two extension tools for enhancing the compression performance of prediction-based lossless audio coding are proposed. One is progressive-order prediction of the starting samples at the random access points, where the information of previous samples is not available. The first sample is coded as is, the second is predicted by first-order prediction, the third is predicted by second-order prediction, and so on. This can be efficiently carried out with PAR-COR (PARtial autoCORrelation) coefficients. The second tool is interchannel joint coding. Both predictive coefficients and prediction error signals are efficiently coded by interchannel differential or three-tap adaptive prediction. These new prediction tools lead to a steady reduction in bit rate when random access is activated and the interchannel correlation is strong.

关键词： audio coding Linear predictive coding Decoding Performance loss Bit rate Entropy Autocorrelation Multiple signal classification Propagation losses MPEG standards

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MPEG-4 ALS: an emerging standard for lossless audio coding

MPEG-4 ALS: an emerging standard for lossless audio coding

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Data Compression Conference (DCC)

作者： T. Liebchen Y.A. Reznik Technical University Berlin Berlin Germany Real NEtworks Inc. Seattle WA USA

This paper provides a brief overview of an emerging ISO/IEC standard for lossless audio coding, MPEG-4 ALS and explains the choice of algorithms used in its design, and compare it to current state-of-the-art algorithm... 详细信息

关键词： MPEG 4 Standard audio coding Data compression

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Improved lossless audio coding using the noise-shaped IntMDCT

Improved lossless audio coding using the noise-shaped IntMDC...

引用

IEEE Digital Signal Processing Workshop

作者： Y. Yokotani R. Geiger G. Schuller S. Oraintara K.R. Rao Electrical Engineering Department University of Texas Arlington Arlington TX USA Fraunhofer Institute for Digital Media Technology Ilmenau Germany

This paper discusses approximation noise shaping to improve the efficiency of the integer modified discrete cosine transform (IntMDCT)-based lossless audio codec. The scheme is applied to rounding operations associated with lifting steps to shape the noise spectrum towards the low frequency bands. In this paper, constraints on the noise shaping filter and a design procedure with the constraints are discussed. Several noise shaping filters are designed and experimental results showing the improvement are presented.

关键词： audio coding Noise shaping Low-frequency noise Frequency Noise cancellation Filters Roundoff errors Floors Noise level Discrete cosine transforms

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Fine-granular scalable and error resilient audio coding by tree-structured quantization

Fine-granular scalable and error resilient audio coding by t...

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

作者： S. Johansen A. Perkis T.A. Ramstad A.S. Bopardikar Quantifiable Quality of Service in Communications Systems Norwegian University of Science and Technology Trondheim Norway Department of Telecommunications Norwegian University of Science and Technology Trondheim Norway

This paper presents a scheme for scalable, error resilient coding of audio signals. The scheme is implemented here as part of a subband-based audio codec and involves the union of two interconnected ideas: tree-structured vector quantization (TSVQ) and reallocation of bits (RoB). The former ensures that the encoding procedure results in fixed-length codes while the latter is a low-complexity scaling algorithm that computes optimal bitrates for each subband when the bitstream needs to be downscaled to a lower bitrate. The functionality enabled by these two systems assumes great importance when the encoded material is to be delivered over channels and networks that have a time-varying bitrate, such as IP-based wireless networks. Such channels are often error-prone and the potentially debilitating effects of introduced errors are mitigated by the inherent fixed-length property of all codewords. We present results that highlight the performance of this system under noisy conditions and show that the degradation in quality is graceful.

关键词： audio coding Bit rate Filter bank Codecs Quality of service Paper technology Vector quantization Resilience Signal resolution Time varying systems

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A scalable speech and audio coding scheme with continuous bitrate flexibility

A scalable speech and audio coding scheme with continuous bi...

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

作者： B. Kovesi D. Massaloux A. Sollaud France Telecom Research and Development Center Lannion France

Networks are getting more and more heterogeneous. Scalable codecs are especially suited for such a context as they permit the bitrate to be lowered in a simple way, at any point of the transmission, for adaptation to network conditions and to terminal capacities. Classically, scalable codecs are organised in layers and scalability is obtained by sending more or fewer layers to the decoder. The obtained granularity depends on the layer sizes, and the available bitrates are fixed and limited in number. The paper presents a novel scalable audio coding scheme where the bitrates vary continuously between a minimal and a maximal value, allowing free modification of the bitrate. With this novel approach, all bitrates are valid, sending even one more bit results in different output signal with statistically growing quality. Test results show that this method provides quality as good as or even better than that of a non-scalable version.

关键词： Speech coding audio coding Bit rate Codecs Decoding Bandwidth Scalability Testing Wideband MPEG 4 Standard

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Improved implementation of MDCT in MP3 audio coding

Improved implementation of MDCT in MP3 audio coding

引用

Asia-Pacific Conference on Communications

作者： V. Nikolajevic G. Fettweis Vodafone Chair Mobile Communications Systems Dresden University of Technology Dresden Germany

Forward and inverse modified discrete cosine transform (MDCT) are two of the most computational intensive operations in the MPEG audio coding standard. So, their efficient implementations are very important. We present an improved algorithm for fast implementation of forward and inverse MDCT for layer III in MPEG-1 and MPEG-2 audio coding standards. The proposed algorithm requires fewer operations than recently proposed one. Furthermore, it needs one step less in signal flow graph and it employs smaller shuffle block compared to recently proposed algorithm.

关键词： Digital audio players audio coding Forward contracts Flow graphs Decoding Mobile communication Discrete cosine transforms Hardware Code standards Codecs

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Architecture design of MDCT-based psychoacoustic model co-processor in MPEG advanced audio coding

Architecture design of MDCT-based psychoacoustic model co-pr...

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Tsung-Han Tsai Shih-Way Huang Yi-Wen Wang Department of Electrical Engineering National Central University Taiwan Department of Electrical Engineering National Taiwan University Taipei Taiwan

ISBN: (纸本)078038251X

A novel architecture of MDCT-based Psychoacoustic Model (PAM) in an MPEG-2/4 Advanced audio coding (AAC) encoder is proposed. PAM, a very important role in AAC encoding, not only determines sound quality of a given encoder but also influences a lot in computational complexity. Our paper presents a dedicated hardware design, which is different from traditional programmable approaches. Moreover, to reduce the complexity, we replace the calculations of spreading function with reduced fixed coefficients, and decrease the transform kernels from three to two units.

关键词： Psychoacoustic models Coprocessors audio coding Kernel Encoding Hardware Masking threshold Computer architecture Noise shaping Signal to noise ratio

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