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检索条件"主题词=Audio coding"

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Principles and analysis of the squeezing approach to low bit rate spatial audio coding

Principles and analysis of the squeezing approach to low bit...

引用

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

作者： Cheng, Bin Ritz, Christian Burnett, Ian Univ Wollongong Whisper Labs Wollongong NSW 2500 Australia

ISBN: (纸本)9781424407286

This paper presents a novel solution to multichannel spatial audio coding: Spatial Squeezing Surround audio coding (S(3)AC). The S(3)AC scheme analyses a multichannel audio signal and downmixes it into a stereo signal pair containing both the monophonic properties of audio sources and their localization information;this avoids the need for side information. The approach uses time-frequency analysis of a spatial audio scene and exploits virtual sources and amplitude panning techniques to 'squeeze' 360 degrees of a horizontal soundfield to a 60 degrees stereo signal pair. In comparison with other spatial audio coding techniques, S(3)AC significantly advances in-band encoding of the localization information in the original sound scene and achieves accurate recoverability of dynamic localized sources.

关键词： audio coding audio systems

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Dynamic algorithm transforms for reconfigurable real-time audio coding processor

Dynamic algorithm transforms for reconfigurable real-time au...

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International Conference on Parallel Computing in Electrical Engineering (PAR ELEC 2002)

作者： Petrovsky, A Petrovsky, A State Univ Informat & Radioelect Comp Sci Dept Belaruasian Minsk 220027 BELARUS

ISBN: (纸本)0769517307;0769517315

In this paper, dynamic algorithm transforms (DTA) for reconfigurable real-time processor for audio coding based on the adaptive wavelet packet (WP) decomposition are presented DAT techniques is to constrain a minimum cost subband decomposition of wavelet transform by maximizing the minimum masking threshold (which is limited by the perceptual entropy) in every subband for the given embedded processor architecture and temporal resolution.

关键词： audio coding Digital signal processing Electronic mail Entropy Heuristic algorithms Real time systems Signal processing Signal processing algorithms Wavelet packets Wavelet transforms

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A Window Switching Algorithm for AVS audio coding

A Window Switching Algorithm for AVS Audio Coding

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3rd International Conference on Wireless Communications, Networking and Mobile Computing (WiCOM 2007)

作者： Chen Shuixian Ai Haojun Hu Ruimin Deng Guiping Wuhan Univ Natl Engn Res Ctr Multimedia Software Key Lab Multimedia & Network Commun Engn Comp Sch Wuhan 430072 Hubei Peoples R China

ISBN: (纸本)9781424413119

In this paper, we present a window switching algorithm employed in A VS (audio Video coding Standard of China) generic audio coding. The algorithm - Energy and Unpredictability Measure based Window Switching Decision (ENUPM-WSD) - achieves high performance to complexity ratio by taking the advantages of both the low complexity 0 energy based decision in time domain and the high accuracy of unpredictability based decision in frequency domain. It significantly improves encoded audio quality, especially those with large transient portions, while preserves low average computational complexity in A VS generic audio encoder. Due to the merits, it is accepted as a recommendation module of the A VS audio standard.

关键词： window switching transient detection audio coding IntMDCT AVS audio

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TEMPORAL NOISE SHAPING ON MDCT SUBBAND SIGNALS FOR TRANSFORM audio coding

TEMPORAL NOISE SHAPING ON MDCT SUBBAND SIGNALS FOR TRANSFORM...

引用

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

作者： Fueg, Richard Edler, Bernd Int Audio Labs Erlangen Wolfsmantel 33 D-91058 Erlangen Germany

ISBN: (纸本)9798350323726

A common problem when coding transient signals in transform audio codecs is the temporal unmasking of quantization noise leading to artifacts such as pre-echoes. In this work, a new method for temporal shaping of quantization noise is introduced. Subband merging/splitting with overlapping windows is applied on the MDCT (Modified Discrete Cosine Transform) spectrum in the encoder and decoder. This allows direct modification of the temporal subband signals for different frequency ranges in a way that results in a flattening of the temporal envelopes in the encoder before quantization and a reshaping to their original envelope at the decoder. The reshaping process at the decoder not only restores the envelopes of the original subband signals, but also shapes the quantization noise accordingly, which results in quantization noise temporally shaped similarly to the original subband signal. This avoids temporal unmasking of quantization noise and the pre-echo artifact. For the temporal flattening/reshaping process the application of gains transmitted via Autoregressive model as well as a low side information approach via companding are proposed and evaluated regarding their noise shaping capabilities in an experimental codec system.

关键词： audio coding mdct temporal noise shaping (tns) subband merging/splitting

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An MDCT domain frame-loss concealment technique for MPEG advanced audio coding

An MDCT domain frame-loss concealment technique for MPEG adv...

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

作者： Ryu, Sang-Uk Rose, Kenneth Univ Calif Santa Barbara Dept Elect & Comp Engn Santa Barbara CA 93106 USA

ISBN: (纸本)9781424407286

We propose a frame loss concealment technique for decoders compatible with MPEG advanced audio coding (AAC). The spectral information of the lost frame is estimated in the modified discrete cosine transform (MDCT) domain via efficient techniques that are tailored to individual source signal components: In noise-like spectral bins the MDCT coefficients are obtained by shaped-noise insertion, while coefficients in tone-dominant bins are estimated by frame interpolation followed by a refinement procedure so as to optimize the fit of the concealed frames with neighboring frames. Experimental results demonstrate that the proposed technique offers performance superior to techniques adopted in commercial AAC decoders.

关键词： audio coding AAC frame loss concealment noise insertion prediction

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Scalable embedded zero tree wavelet packet audio coding 3

Scalable embedded zero tree wavelet packet audio coding

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3rd IEEE Workshop on Signal Processing Advances in Wireless Communications (SPAWC 01)

作者： Chang, PC Lin, JH Natl Cent Univ Dept Elect Engn Chungli 32045 Taiwan

ISBN: (纸本)0780367200

Multimedia transmission over Internet is getting popular and increasingly important;In particular, scalable coding is desirable for heterogeneous network with varies bandwidths. In this work, we propose a scalable embedded zero tree wavelet packet (Scalable EZWP) audio coding system that is a scalable audio compression system using wavelet packet decomposition and embedded zero-tree coding. We focus on multi-layer low bitrate coding which delivers high perceptual quality. In the base layer, the overlapped audio segment is first transformed by wavelet packet. Then the local significant coefficients are extracted, quantized, and coded by variable length coding. In the enhancement layer and the full band layer, the residual signal that is the difference between the original and the output of the previous layer is coded via EZW with psychoacoustic model and arithmetic coding. The target bit rates for three layers are 16, 32, and 64 1Kbps, respectively. The performance of the proposed coding system is only slightly inferior to MPEG-1 layer 3 at 64 Kbps while it provides bitrate scalability that is suitable for multimedia distribution over Internet heterogenous networks.

关键词： Arithmetic audio coding audio compression Bandwidth Bit rate Internet Nonhomogeneous media Psychoacoustic models Scalability Wavelet packets

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Research on Asymmetry Multi-terminal sources audio coding Algorithm

Research on Asymmetry Multi-terminal sources Audio Coding Al...

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International Conference on Communication Software and Networks

作者： Jiang Yan Ning Gengxin Wei Gang Yang Yucun S China Univ Technol Sch Elect & Informat Engn Guangzhou Guangdong Peoples R China

ISBN: (纸本)9780769535227

Multi-terminal sources coding refers to separate lossy encoding and joint decoding of two or more correlated sources. Based on good output performance it can effectively reduce encoding complexity. With focus on the asymmetry case, This paper designs a asymmetry multi-terminal sources audio coding algorithm, then analyses and simulates it. The encouraging simulation results show multi-terminal sources audio coding is feasible, simple and can get higher acoustical effect.

关键词： Multi-terminal sources coding Distributed coding audio coding

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A new psychoacoustical masking model for audio coding applications

A new psychoacoustical masking model for audio coding applic...

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： van de Par, S Kohlrausch, A Charestan, G Heusdens, R Philips Res Labs Digital Signal Proc Grp NL-5656 AA Eindhoven Netherlands

ISBN: (纸本)0780374029

The use of psychoacoustical. masking models for audio coding applications has been wide spread over the past decades. In such applications, it is typically assumed that the original input signal serves as a masker for the distortions that are introduced by the lossy coding method that is used. Such masking models are based on the peripheral bandpass filtering properties of the auditory system and basically evaluate the distortion-to-masker ratio within each auditory filter. Up to now these models have been based on the assumption that the masking of distortions is governed by the auditory filter for which the ratio between distortion and masker is largest. This assumption, however, is not in line with some new findings within the field of psychoacoustics. A more accurate assumption would be that the human auditory system is able to integrate distortions that are present within a range of auditory filters, In this contribution a new model is presented which is in line with new psychoacoustical studies and which is suitable for application within an audio codec. Although this model can be used to derive a masking curve, the model also gives a measure for the detectability of distortions provided that distortions are not too large.

关键词： audio coding peripheral bandpass distortion masking input signal psychoacoustical masking model acoustic signal processing band-pass filters channel bank filters calibration codecs filtering theory hearing data compression

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ULTRA-LOW LATENCY audio coding BASED ON DPCM AND BLOCK COMPANDING

ULTRA-LOW LATENCY AUDIO CODING BASED ON DPCM AND BLOCK COMPA...

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14th International Workshop on Image Analysis for Multimedia Interactive Services (WIAMIS)

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

ISBN: (纸本)9781479908332

A low delay audio coding scheme with good perceptual audio quality for a desired limited bit rate is presented. The proposed audio coding scheme is based on differential pulse code modulation (DPCM) and block companded (BC) quantization. Prediction is realized as a FIR filter in lattice structure. DPCM performs in feedback manner, therefore no transmission of prediction filter coefficients is needed. The incorporation of BC quantization in the DPCM relies on a prediction error recalculation scheme. The use of BC quantization in the DPCM allows to accurately follow the prediction error signal. This improves the perceptual audio quality significantly compared to a plain DPCM with an adaptive quantizer. An algorithmic delay below a half millisecond and an overhead of less than a half bit per sample is introduced due to the short fixed block length of the BC quantizer. Therefore, a real time bidirectional audio application is achievable.

关键词： audio coding linear predictive coding block compander

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Low complexity rate-distortion optimized time-segmentation for audio coding

Low complexity rate-distortion optimized time-segmentation f...

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Workshop on Applications of Signal Processing to audio and Acoustics

作者： Rodbro, CA Christensen, MG Nordén, F Jensen, SH Aalborg Univ Dept Commun Technol DK-9220 Aalborg O Denmark

ISBN: (纸本)0780391543

In this paper, we investigate it reduced complexity approach to rate-distortion optimized time-segmentation in audio coding. Instead of the conventional closed-loop approach for determining the coding distortions, they are estimated from a set of features extracted from the audio signal. Care is taken to ensure that properties such as convex and non-increasing rate-distortion curves carry over front the training data to the estimated rate-distortion pairs. With computational complexity reductions of a factor close to 10, perceptual listening tests reveal a slight reduction of the signal quality, while maintaining a large improvement over fixed segmentation.

关键词： audio coding computational complexity rate distortion theory coding distortions computational complexity reductions features extraction low complexity rate-distortion rate-distortion time-segmentation

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