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检索条件"主题词=Audio coding"

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An analysis of perceptual artifacts in MPEG scalable audio coding

An analysis of perceptual artifacts in MPEG scalable audio c...

引用

Data Compression Conference (DCC 2002)

作者： Creusere, CD New Mexico State Univ Klipsch Sch Elect & Comp Engn Las Cruces NM 88003 USA

ISBN: (纸本)0769514774

In this paper we study coding artifacts in MPEG-compressed scalable audio. Specifically, we consider the MPEG advanced audio coder (AAC) using bit slice scalable arithmetic coding (BSAC) as implemented in the MPEG 4 reference software. First, we perform human subjective testing using the comparison category rating (CCR) approach, quantitatively comparing the performance of scalable BSAC with the nonscalable TwinVQ and AAC algorithms. This testing indicates that scalable BSAC performs very poorly relative to TwinVQ at the lowest bitrate considered (16 kb/s) largely because of an annoying and seemingly random mid-range tonal signal that is superimposed onto the desired output. In order to better understand and perceptually quantify the various forms of distortion introduced into compressed audio at low bit rates, we apply two analysis techniques: Reng probing and time-frequency decomposition. The Reng probing technique is capable of separating the Linear time-invariant component of a multirate system from its nonlinear and periodically time-varying components. Using this technique, we conclude that aliasing is probably not the cause of the annoying tonal signal;instead, time-frequency analysis indicates that its cause is most likely suboptimal bit allocation.

关键词： Arithmetic audio coding Bit rate Humans MPEG 4 Standard Nonlinear distortion Performance evaluation Testing Time frequency analysis Time varying systems

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Multiple Description audio coding for Wireless Low-Frequency Sound Zones

Multiple Description Audio Coding for Wireless Low-Frequency...

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Data Compression Conference (DCC)

作者： Ostergaard, Jan Pedersen, Christian Zhou, Mo de Koeijer, Niels Moller, Martin Aalborg Univ Fredrik Bajers Vej 7B DK-9000 Aalborg Denmark Bang & Olufsen Bang Olufsen Alle 1 DK-7600 Struer Denmark

ISBN: (纸本)9798350347951

We present a joint design of sound zone control filters and robust audio coding for wireless low frequency sound zones. The audio signal is filtered using sound zone control filters and encoded using a multiple-description coder. The control filters and the multiple-description coder are combined in a nested loop. The inner loop performs filtering for sound zone control and generates multiple descriptions using oversampling and closed-loop prediction. The outer loop performs noise shaping and guarantees a trade-off between robustness and quality of the descriptions. A closed-form expression for the optimal sound-zone control filters are provided, and a simulation study demonstrates that even at moderate packet loss rates, a significant gain is possible compared to not using multiple descriptions.

关键词： audio coding Multiple description coding sound zones wireless streaming

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IMPROVED PHASE PARAMETER ANALYSIS AND SYNTHESIS FOR PARAMETRIC STEREO audio coding

IMPROVED PHASE PARAMETER ANALYSIS AND SYNTHESIS FOR PARAMETR...

引用

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

作者： Hyun, Dong-il Seo, Jeongil Park, Young-cheol Youn, Dae Hee Yonsei Univ Dept Elect & Elect Eng Seoul 120749 South Korea

ISBN: (纸本)9781457705397

This paper proposes an improved phase parameter anlysis and synthesis method for parametric stereo audio coding. For stereo signal with low interchannel coherence (IC), conventional phase parameter analysis and synthesis methods show poor quality, since they overlook the effect of IC to the phase parameters defined. The proposed system defines new phase parameters refleeting IC information, so that we can alleviate the deterioration of audio quality due to the synthesis of excessive phase difference, especially when IC is low. Subjective listening test results are presented to show that the proposed method provides always better audio quality than the conventional methods.

关键词： interchannel phase difference interchannel coherence parametric stereo audio coding

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SPARSE DECOMPOSITION OF audio SIGNALS USING A PERCEPTUAL MEASURE OF DISTORTION. APPLICATION TO LOSSY audio coding 18

SPARSE DECOMPOSITION OF AUDIO SIGNALS USING A PERCEPTUAL MEA...

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18th International Conference on Digital audio Effects (DAFx)

作者： Toumi, Ichrak Derrien, Olivier CNRS LMA 31 Chemin Joseph Aiguier F-13402 Marseille 20 France Univ Toulon & Var F-13402 Marseille 20 France

State-of the art audio codecs use time-frequency transforms derived from cosine bases, followed by a quantification stage. The quantization steps are set according to perceptual considerations. In the last decade, several studies applied adaptive sparse time-frequency transforms to audio coding, e.g. on unions of cosine bases using a Matching-Pursuit-derived algorithm [1]. This was shown to significantly improve the coding efficiency. We propose another approach based on a variational algorithm, i.e. the optimization of a cost function taking into account both a perceptual distortion measure derived form a hearing model and a sparsity constraint, which favors the coding efficiency. In this early version, we show that, using a coding scheme without perceptual control of quantization, our method outperforms a codec from the literature with the same quantization scheme [1]. In future work, a more sophisticated quantization scheme would probably allow our method to challenge standard codecs e.g. AAC.

关键词： audio coding Sparse approximation Iterative thresholding algorithm Perceptual model

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Quantized sparse approximation with iterative thresholding for audio coding

Quantized sparse approximation with iterative thresholding f...

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32nd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Yaghoobi, M. Blumensath, T. Davies, M. Univ Edinburgh Joint Res Inst Signal & Image Proc Inst Digital Commun Edinburgh EH8 9YL Midlothian Scotland

ISBN: (纸本)9781424407286

Sparse coding is a new field in signal processing with possible applications to source coding. In this paper we present a new method that combines the problems of sparse signal approximation with coefficient quantization. This method uses over-complete dictionaries and exploits signal redundancy. The proposed method will be derived as an extension of a recently presented method (iterative thresholding) to find sparse representations of signals. Because in digital communication and storage we need a quantized representation of the signal, instead of quantization of sparse representations a posteriori, we propose a refined method that combines sparse approximation and quantization. To compare the proposed method to a posteriori quantization, we present an audio example.

关键词： sparse approximation quantization iterative thresholding audio coding signal representation

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PREDICTIVE audio coding USING RATE-DISTORTION-OPTIMAL PRE- AND POST-FILTERING

PREDICTIVE AUDIO CODING USING RATE-DISTORTION-OPTIMAL PRE- A...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics (WASPAA)

作者： Moussa, Obada Alhaj Li, Minyue Kleijn, W. Bastiaan KTH Royal Inst Technol Stockholm Sweden

ISBN: (纸本)9781457706936

A natural approach to audio coding is to use a rate-distortion optimal design combined with a perceptual model. While this approach is common in transform coding, existing predictive-coding based audio coders are generally not optimal and they benefit from heuristically motivated post-filtering. As delay requirements often force the use of predictive coding, we consider audio coding with a pre- and post-filtered predictive structure that was recently proven to be asymptotically optimal in the rate-distortion sense [1]. We show that this audio coding is efficient in achieving the state-of-the-art performance. We also show that the pre- filter plays a relatively minor role. This leads to an analytic approach for optimizing the post-filter and the predictor at each rate, eliminating the need for manual re-tuning whenever a different rate is called for. In a subjective test, the theoretically optimized post-filter provided a better performance than a conventional post-filter.

关键词： audio coding Noise Quantization Rate-distortion Resource description framework Speech audio coders audio coding filtering theory optimisation optimization approach post-filtered predictive structure predictive audio coding rate-distortion optimal design r

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A comparison of differential schemes for low-rate sinusoidal audio coding

A comparison of differential schemes for low-rate sinusoidal...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Jensen, J Heusdens, R Delft Univ Technol Dept Mediamat NL-2600 AA Delft Netherlands

ISBN: (纸本)0780378504

Sinusoidal coding plays an important in low rate audio coding. Typically, differential techniques are employed to reduce the bit rate for representing the sinusoidal parameters. In this paper we compare several schemes for time-differential (TD) and frequency-differential (FD) representation of the sinusoidal model parameters. We show through simulation experiments that bit rates obtained with FD techniques are competitive with those achieved using TD techniques, provided that a variable length segmentation algorithm is applied to the input signal first. This result is important because FD techniques (in contrast to TD techniques) do not rely on the presence of previous segments for correct decoding, and therefore offer robust performance in a lossy packet channel environment.

关键词： audio coding audio compression Bit rate Decoding Encoding Frequency Noise robustness Performance loss Signal processing algorithms Speech codecs

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An Embedded Stereo Speech and audio coding Method Based on Principal Component Analysis

An Embedded Stereo Speech and Audio Coding Method Based on P...

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IEEE International Symposium on Signal Processing and Information Technology (ISSPIT)

作者： Jia, Mao-shen Bao, Chang-chun Liu, Xin Li, Xiao-ming Li, Ru-wei Beijing Univ Technol Speech & Audio Signal Proc Lab Sch Elect Informat & Control Engn Beijing 100124 Peoples R China

ISBN: (纸本)9781467307536

In this paper a compressive sampling method of MLT coefficients which is used for extracting stereo information is adopted based on principal component analysis (PCA) and Modulated Lapped Transform (MLT). With this method, an embedded variable bit-rates stereo speech and audio coding algorithm is proposed in this paper. In this codec, the stereo signal sampled at 32 kHz and 16 kHz can be coded in terms of scalable bit rates, the structure of bit-stream is embedded and the bit-stream can be divided into several layers. The core codec is ITU-T G.729.1 which can process mono signal with 7 kHz bandwidth. Besides there are four extra bit-rates added include 40, 48, 56, and 64kb/s. The maximum bit-rates of wide band stereo signal and super-wideband stereo signal are 48kb/s and 64kb/s, respectively. The objective and subjective test results show that the quality of the proposed codec is no worse than the reference codec which is requested by ITU-T

关键词： speech coding audio coding stereo coding embedded coding

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Ultra-low delay lossy audio coding using DPCM and block companded quantization

Ultra-low delay lossy audio coding using DPCM and block comp...

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Australian Communications Theory Workshop (AusCTW)

作者： Simkus, Gediminas Holters, Martin Zoelzer, Udo Univ Hamburg Univ Fed Armed Forces Dept Signal Proc & Commun Hamburg Germany

ISBN: (纸本)9781467346757;9781467346733

Real-time audio transmission requires good quality for a restricted channel capacity and minimum latency. An ultra-low delay audio coding scheme based on differential pulse code modulation (DPCM) and block companded quantization is presented. The prediction filter of the base backward DPCM codec is attained as a FIR filter in lattice structure. The proposed block based audio coding introduces an algorithmic latency below one millisecond and an overhead of less than a half bit per sample. The advantage of the block companded quantization with DPCM is the capability to follow rapid changes in the signal. Therefore, it significantly improves the perceptual audio quality compared to a plain DPCM coding scheme with an adaptive quantizer.

关键词： FIR filters audio coding channel capacity codecs differential pulse code modulation quantisation (signal)

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A new sinusoidal modelling approach for parametric speech and audio coding 3

A new sinusoidal modelling approach for parametric speech an...

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3rd International Symposium on Image and Signal Processing and Analysis (ISPA)

作者： Vera-Candeas, P Ruiz-Reyes, N Curpián-Alonso, J Rosa-Zurera, M Univ Jaen Polytech Sch Dept Elect Jaen 23700 Spain

ISBN: (纸本)953184061X

In this paper we propose a new approach based on energy-adaptive matching pursuits to improve sinusoidal modelling for parametric speech and audio coding. To reduce the complexity of the algorithm, an over-complete dictionary composed of complex exponentials is used. An analysis-synthesis windows scheme that avoids overlapping is proposed. For efficient quantization of sinusoidal model parameters, a new algorithm that largely reduces the side information required by the decoder is described. Experimental results show evidence of the advantages of the proposed method to be integrated into multi-parts models for parametric speech/audio coding.

关键词： audio coding Decoding Dictionaries Matching pursuit algorithms Parameter estimation Psychoacoustic models Pursuit algorithms Signal analysis Signal processing algorithms Speech coding

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