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Practical design of delta-sigma multiple description audio coding

EURASIP JOURNAL ON audio SPEECH AND MUSIC PROCESSING 2014年第1期2014卷 1页

作者： Leegaard, Jack Ostergaard, Jan Jensen, Soren Holdt Zamir, Ram Aalborg Univ Dept Elect Syst DK-9220 Aalborg Denmark Tel Aviv Univ Dept Elect Engn Syst IL-69978 Tel Aviv Israel

It was recently shown that delta-sigma quantization (DSQ) can be used for optimal multiple description (MD) coding of Gaussian sources. The DSQ scheme combined oversampling, prediction, and noise-shaping in order to trade off side distortion for central distortion in MD coding. It was shown that asymptotically in the dimensions of the resampling, prediction, and noise-shaping filters as well as asymptotically in the quantizer dimensions, all rate-distortion points on the symmetric quadratic Gaussian MD rate-distortion function could be achieved. In this work, we show that this somewhat theoretical framework is suitable for practical low-delay MD audio coding. In particular, we design a practical MD audio coder with two descriptions and provide simulations on real audio data. The simulations demonstrate that even when using low-dimensional noise-shaping, prediction, and resampling filters, it is possible to obtain good quality audio in the presence of packet losses. Simulations on real audio reveal that, contrary to existing designs, it is straightforward to obtain a large number of trade-off points between side distortion and central distortion, which makes the proposed coder suitable for a wide range of applications.

关键词： audio coding Multiple descriptions Noise shaping Predictive coding

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Efficient audio coding using perfect reconstruction noncausal IIR filter banks

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IEEE TRANSACTIONS ON SPEECH AND audio PROCESSING 1996年第2期4卷 115-123页

作者： Creusere, CD Mitra, SK Weapons Division Naval Air Warfare Center China Lake CA USA Department of Electrical and Computer Engineering University of California Santa Barbara CA USA

In this paper, we consider the problem of implementing perfect reconstruction modulated polyphase filter banks having infinite impulse response (IIR) subfilters in systems that must process infinite length 1-D inputs. This task is complicated by the fact that the synthesis bank needed to ensure perfect reconstruction in such a system generally has noncausal or reverse-time filter sections. To overcome this limitation, we develop a number of unique buffering schemes that make possible the implementation of such reverse-time filters with infinite length signals. For the important special case when the polyphase subfilters are allpass, we also present a method that allows for very efficient implementation of tree-structured filter banks and wavelet transforms. We then describe a wideband coding algorithm based largely on work done by Lokhoff and Johnston. Using this algorithm, we compare the performance of IIR filter banks with that of more conventional pseudo-QMF and tree-structured QMF banks. The results of these comparisons indicate that one of our filter banks-a tree-structured allpass bank-achieves slightly better performance than equivalent FIR banks in terms of rate and distortion (both objectively and subjectively measured) while requiring lower computational complexity.

关键词： audio coding Filter bank IIR filters Finite impulse response filter Signal synthesis Wavelet transforms Wideband Rate distortion theory Distortion measurement Computational complexity

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Understanding perceptual distortion in MPEG scalable audio coding

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IEEE TRANSACTIONS ON SPEECH AND audio PROCESSING 2005年第3期13卷 422-431页

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

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 nonscaled 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 quantify the distortion introduced into compressed audio at low bit rates, we apply two analysis techniques: Reng bifrequency probing and time-frequency decomposition. Using Reng probing, we conclude that aliasing is most likely not the cause of the annoying tonal signal;instead, time-frequency or spectrogram analysis indicates that its cause is most likely suboptimal bit allocation. Finally, we describe the energy equalization quality metric (EEQM) for predicting the relative perceptual performance of the different coding algorithms and compare its predictive ability with that of ITU Recommendation ITU`-R BS.1387-1.

关键词： audio analysis audio coding audio quality metrics objective quality assessment perceptual distortion scalable coding

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Spatial parameters for audio coding: MDCT domain analysis and synthesis

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MULTIMEDIA TOOLS AND APPLICATIONS 2010年第2期48卷 225-246页

作者： Chen, Shuixian Xiong, Naixue Park, Jong Hyuk Chen, Min Hu, Ruimin Kyungnam Univ Dept Comp Sci & Engn Masan South Korea Wuhan Univ Comp Sch Wuhan 430072 Peoples R China Georgia State Univ Dept Comp Sci Atlanta GA 30303 USA Seoul Natl Univ Sch Engn & Comp Sci Seoul 151744 South Korea

We use Modified Discrete Cosine Transform (MDCT) to analyze and synthesize spatial parameters. MDCT in itself lacks phase information and energy conservation, which are needed by spatial parameters representation. Completing MDCT with Modified Discrete Sine Transform (MDST) into "MDCT-j*MDST" overcomes this and enables the representation in a form similar to that of DFT. And due to overlap-add in time domain, a MDST spectrum can be built perfectly from MDCT spectra of neighboring frames through matrix-vector multiplication. The matrix is heavily diagonal and keeping only a small number of its sub-diagonals is sufficient for approximation. When using MDCT based core coder in spatial audio coding, like Advanced audio coding (AAC), we need no separate transforming for spatial processing, cutting down significantly the computational complexity. Subjective listening tests also show that MDCT domain spatial processing has no quality impairment.

关键词： audio coding MDCT MDST Singular value Spatial parameter

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Efficient bit-rate scalability for weighted squared error optimization in audio coding

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IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2006年第4期14卷 1313-1327页

作者： Aggarwal, Ashish Regunathan, Shankar L. Rose, Kenneth Harman Consumer Grp Northridge CA 91329 USA Microsoft Corp Redmond WA 98052 USA Univ Calif Santa Barbara Dept Elect & Comp Engn Santa Barbara CA 93106 USA

We propose two quantization techniques for improving the bit-rate scalability of compression systems that optimize a weighted squared error (WSE) distortion metric. We show that quantization of the base-layer reconstruction error using entropy-coded scalar quantizers is suboptimal for the WSE metric. By considering the compandor representation of the quantizer, we demonstrate that asymptotic (high resolution) optimal scalability in the operational rate-distortion sense is achievable by quantizing the reconstruction error in the compandor's companded domain. We then fundamentally extend this work to the low-rate case by the use of enhancement-layer quantization which is conditional on the base-layer information. In the practically important case that the source is well modeled as a Laplacian process, we show that such conditional coding is implementable by only two distinct switchable quantizers. Conditional coding leads to substantial improvement over the companded scalable quantization scheme introduced in the first part, which itself significantly outperforms standard techniques. Simulation results are presented for synthetic memoryless Laplacian sources with P-law companding, and for real-world audio signals in conjunction with MPEG AAC. Using the objective noise-mask ratio (NMR) metric, the proposed approaches were found to result in bit-rate savings of a factor of 2 to 3 when implemented within the scalable MPEG AAC. Moreover, the four-layer scalable coder consisting of 16-kb/s layers achieves performance close to that of the 64-kb/s nonscalable coder on the standard test database of 44.1-kHz audio.

关键词： AAC audio coding bit-rate scalability embedded transmission quantization

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Multiple description perceptual audio coding with correlating transforms

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IEEE TRANSACTIONS ON SPEECH AND audio PROCESSING 2000年第2期8卷 140-145页

作者： Arean, R Kovacevic, J Goyal, VK Bell Labs Lucent Technol Murray Hill NJ 07974 USA

In audio communication over a lossy packet network, concealment techniques are used to mitigate the effects of lost packets. This concealment is markedly improved if the compressed representation retains redundancy to aid in the estimation of lost information. A perceptual audio coder employing multiple description correlating transforms demonstrates this phenomenon.

关键词： audio coding multiple descriptions packetized audio robust communication

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Fast Subband Filtering in MPEG audio coding

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IEEE SIGNAL PROCESSING LETTERS 1994年第2期1卷 26-28页

作者： Konstantinides, Konstantinos Hewlett Packard Labs Palo Alto CA 94303 USA

Subband filtering is one of the most compute-intensive operations in the MPEG audio coding standard. In this paper, we prove that the matrixing operations in MPEG subband filtering can be efficiently computed using fa... 详细信息

关键词： discrete cosine transforms Moving Pictures Experts Group audio coding subband Filtration

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Regressive implementations for the forward and inverse MDCT in MPEG audio coding

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IEEE SIGNAL PROCESSING LETTERS 1996年第4期3卷 116-118页

作者： Chiang, HC Liu, JC Department of Electrical Engineering Tatung Institute of Technology Taipei Taiwan

Modified discrete cosine transform (MDCT) and inverse modified discrete cosine transform (IMDCT) are two of the most compute-intensive operations in layer III of the MPEG audio coding standard, In this letter, we first derive two sinusoidal recursive formulas for the transform kernels of the MDCT and IMDCT. Then we demonstrate that general length of MDCT and IMDCT can be efficiently implemented by using the regressive structure derived from the sinusoidal recursive formulas, The proposed regular structure is particularly suitable for parallel VLSI realization, Besides, it also provides an efficient on-line computation scheme and is especially suitable for the computation of variable length MDCT and IMDCT when involving dynamical window switching in MPEG audio coding standard.

关键词： audio coding Very large scale integration Hardware

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Quantisation noise control in perceptual audio coding using low selectivity filter banks

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ELECTRONICS LETTERS 2002年第16期38卷 932-933页

作者： Martínez-Muñoz, D Rosa-Zurera, M Cruz-Roldán, F López-Ferreras, F Ruiz-Reyes, N Univ Jaen Escuela Univ Politecn Dept Elect Jaen 23700 Spain Univ Alcala de Henares Escuela Politecn Dept Teoria Senal & Comunicac Madrid 28871 Spain

The problem of computing, in a subband audio coder, the maximum quantisation noise power that can be injected in each band to ensure transparent coding when low selectivity filter banks are used, is addressed. A low complexity strategy, taking into account the frequency responses of the synthesis filter bank, is proposed for achieving an overall distortion due to quantisation noise always below the masking threshold (provided by a psycho-acoustic model) for any length prototype filters.

关键词： quantisation (signal) frequency response ISO/MPEG-1 standard low complexity strategy maximum quantisation noise power filtering theory frequency responses masking threshold Codes quadrature mirror filters quantisation noise control perceptual audio coding subband audio coder audio coding white noise transparent coding psychoacoustic model Filtering methods in signal processing low selectivity filter banks Speech and audio coding Signal processing and conditioning equipment and techniques

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HHT-based audio coding

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SIGNAL IMAGE AND VIDEO PROCESSING 2015年第1期9卷 107-115页

作者： Khaldi, Kais Boudraa, Abdel-Ouahab Torresani, Bruno Chonavel, Thierry BCRM Brest Ecole Navale IRENav F-29240 Brest 9 France U2S ENIT Tunis Tunisia Univ Aix Marseille 1 CMI LATP F-13453 Marseille 13 France LabSTICC UMR F-29238 Brest 3 France

In this paper, a new audio coding scheme combining the Hilbert transform and the empirical mode decomposition (EMD) is introduced. Based on the EMD, the coding is fully a data-driven approach. audio signal is first decomposed adaptively, by EMD, into intrinsic oscillatory components called intrinsic mode functions (IMFs). The key idea of this work is to code both instantaneous amplitude (IA) and instantaneous frequency (IF), of the extracted IMFs, calculated using Hilbert transform. Since IA (resp. IF) is strongly correlated, it is encoded via a linear prediction technique. The decoder recovers the original signal by superposition of the demodulated IMFs. The proposed approach is applied to audio signals, and the results are compared to those obtained by advanced audio coding (AAC) and MP3 codecs, and wavelets-based compression. coding performances are evaluated using the bit rate, objective difference grade (ODG) and noise to mask ratio (NMR) measures. Based on the analyzed audio signals, overall, our coding scheme performs better than wavelet compression, AAC and MP3 codecs. Results also show that this new scheme has good coding performances without significant perceptual distortion, resulting in an ODG in range [-1, 0] and large negative NMR values.

关键词： audio coding Empirical mode decomposition Intrinsic mode function Hilbert transform Hilbert-Huang transform Linear prediction

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