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Improved audio coding using a psychoacoustic model based on a cochlear filter bank

IEEE TRANSACTIONS ON SPEECH AND audio PROCESSING 2002年第7期10卷 495-503页

作者： Baumgarte, F Agere Syst Media Signal Proc Res Dept Berkeley Hts NJ 07922 USA

Perceptual audio coders use an estimated masked threshold for the determination of the maximum permissible just-inaudible noise level introduced by quantization. This estimate is derived from a psychoacoustic model mimicking the properties of masking. Most psychoacoustic models for coding applications use a uniform (equal bandwidth) spectral decomposition as a first step to approximate the frequency selectivity of the human auditory system. However, the equal filter properties of the uniform subbands do not match the nonuniform characteristics of cochlear filters and reduce the precision of psychoacoustic modeling. Even so, uniform filter banks are applied because they are computationally efficient. This paper presents a psychoacoustic model based on an efficient nonuniform cochlear filter bank and a simple masked threshold estimation.. The novel filter-bank structure employs cascaded low-order HR filters and appropriate down-sampling to increase efficiency. The filter responses are. optimized for the modeling of auditory masking effects. Results of the new psychoacoustic model applied to audio coding show better performance in terms of bit rate and/or quality of the new model in comparison with other state-of-the-art models using a uniform spectral decomposition. The low delay of the new model is particularly suitable for low-delay coders.

关键词： audio coding filter bank masked threshold model of masking perceptual model

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Multistream transmission for hybrid IBOC-AM with embedded/multidescriptive audio coding

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IEEE TRANSACTIONS ON BROADCASTING 2002年第3期48卷 179-192页

作者： Lou, HL Sinha, D Sundberg, CEW Bell Labs Lucent Technol Multimedia Commun Res Lab Murray Hill NJ 07974 USA

Hybrid In Band on Channel (IBOC) digital audio broadcasting simultaneously with analog amplitude modulation (AM) has been proposed as a hybrid solution to digital audio broadcasting in the AM band. Since the AM band is crowded and since the available bandwidth per program is limited, adding digital transmission is a challenging proposition. To achieve FM like audio quality, an audio coder rate of 32-64 kb/sec may be required. One of the currently proposed hybrid IBOC-AM systems is 30 kHz wide. Severe second adjacent interference may occur in certain geographical. areas. This may lead to loss of 40% of the effective transmission audio bit rate. For coping with such harsh transmission conditions, we present a solution based on embedded/multidescriptive audio coding with matched multistream transmission in separate frequency bands. With loss of one frequency band, the embedded system blends to a lower audio coder rate with a much better quality than analog AM. The nonembedded system without multistream transmission fails catastrophically when a little more than one sideband is severely interfered with causing a severe discontinuity in quality while blending directly to analog AM. A number of detailed robust embedded systems are outlined. We also show how multistream transmission schemes can be used with nonembedded audio coders. Both daytime and nighttime scenarios are included. This paper contains a catalog of possible systems for different audio quality levels and interference scenarios, including systems with 20 kHz bandwidth rather than 30 kHz.

关键词： amplitude modulation audio coding channel coding digital audio broadcasting error detection coding radio broadcasting

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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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Algorithm for achieving adaptive tiling of time axis for audio coding purposes

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ELECTRONICS LETTERS 2002年第9期38卷 434-435页

作者： Ruiz, N Rosa, M López, F Vera, P Univ Jaen Dept Elect Escuela Politecn Linares Jaen Spain Univ Alcala de Henares Escuela Politecn Dept Teoria Senal & Commun Madrid Spain

A new algorithm for achieving flexible tiling, of the time axis for audio coding purposes is presented, It is based on the calculus of the distances among a predetermined number of time-frequency pairs, From the computed distances. a clustering process determines the final subdivision of each audio frame. Experimental results demonstrates the good performance of the proposed algorithm. which provides high coding, efficiency with a reduced complexity.

关键词： audio coding clustering process audio frame complexity reduction adaptive tiling time-frequency pairs Speech and audio coding time axis high coding efficiency adaptive signal processing

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Efficient DSP architecture for high-quality audio algorithms

Efficient DSP architecture for high-quality audio algorithms

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IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Suk Hyun Yoon Jong Ha Moon Myung Hoon Sunwoo School of Electrical and Computer Engineering Ajou University Suwon South Korea

ISBN: (纸本)0780388348

This paper presents specialized DSP instructions and their hardware architecture for high-quality audio algorithms, such as the MPEG-2/4 advanced audio coding (AAC), Dolby AC-3, MPEG-2 backward compatible (BC), etc. The proposed architecture is specially designed and optimized for the IMDCT (inverse modified discrete cosine transform), and Huffman decoding in the AAC decoding algorithm. Performance comparisons show a significant improvement compared with TMS320C62/spl times/ and ASDSP21060 for the IMDCT computation. Furthermore, the dedicated Huffman accelerator performs the decoding process in only one cycle. The proposed DPU (data processing unit) consists of 107,860 gates and achieves 150 MIPS.

关键词： Digital signal processing Decoding Hardware audio coding Computer architecture Algorithm design and analysis Design optimization Discrete cosine transforms Variable speed drives Data processing

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Design of a high-quality audio-specific DSP core

Design of a high-quality audio-specific DSP core

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IEEE Workshop on Signal Processing Systems (SIPS)

作者： Suk Hyun Yoon Myung Hoon Sunwoo J.H. Moon School of Electrical and Computer Engineering Ajou University Suwon South Korea LG Electronics Inc. Seoul South Korea

This paper proposes a specialized DSP architecture and their instructions, which efficiently support MPEG-2/4 AAC high-quality audio algorithms. The proposed architecture is specially designed and optimized for the IMDCT (inverse modified discrete cosine transform), Huffman decoding, etc. Performance comparisons show significant improvement compared with TMS320C62x and ASDSP21060 for the IMDCT computation. Furthermore, the dedicated Huffman accelerator performs the decoding process in only 2 cycles. The proposed DSP has been synthesized using the Samsung SEC 0.18 /spl mu/m standard cell library. The proposed DSP core consists of 120,283 gates and runs at 200 MHz.

关键词： Digital signal processing Decoding Computer architecture Digital signal processing chips Filter bank Signal processing algorithms Transform coding Computer aided instruction audio coding Quantization

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Comparison of psychoacoustic principles and genetic algorithms in audio compression

Comparison of psychoacoustic principles and genetic algorith...

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IEEE International Conference on Systems Engineering

作者： H. Chen T.L. Yu A Department of Computer Science California State University San Bernardino CA USA

High audio data compression can be achieved by removing irrelevant signal information that is not detectable by even a well-trained or sensitive listener. Contemporary audio coding schemes like MP3, AAC, and Ogg Vorbis identify the irrelevant information during signal analysis by incorporating into the coder several psychoacoustic principles, including absolute hearing thresholds, critical band analysis, simultaneous masking, and temporal masking (Painter and Spanias, 2000). Masking is the process of removing faint but normally audible sound signals that are rendered inaudible as they are very close in frequency to or have much smaller amplitudes than surrounding sounds. Numerous studies have been conducted on genetic algorithms, which solve problems by modeling the Darwinian evolution. The algorithms have been recently applied to audio coding with some success (Galos et al., 2003). To achieve audio compression, genetic algorithms analyze a large number of sound files to determine the chunks that are most likely to contain irrelevant signals. The combination of the irrelevant chunks, form a solution which will be used to compress any sound files. We present in this paper a study of the comparison of applying psychoacoustic principles and genetic algorithms to compress audio signals. We developed a coder to perform the experiment, where like most well-known audio coders, Huffman coding is used to handle lossless compression and modified discrete cosine transform (MDCT) is used to transform the time-domain signals to the frequency domain. The results are compared using signal-to-noise ratios (SNRs) and subjective testing, where eighteen subjects (who are students in CSUSB) are asked to listen and rate the decompressed files by the two methods.

关键词： Psychology Genetic algorithms audio compression Signal analysis audio coding Data compression Digital audio players Auditory system Information analysis Signal processing

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Development and evaluation of an over-sampled wavelet packet audio coder

Development and evaluation of an over-sampled wavelet packet...

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International Symposium on Signal Processing and Its Applications (ISSPA)

作者： T. Surya Gunawan F. Sinaga E. Ambikairajah School of Electrical Engineering and Telecommunications University of New South Wales Sydney NSW Australia

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Stability of the stereo linear prediction schemes

Stability of the stereo linear prediction schemes

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International Symposium on Electronics in Marine (ELMAR)

作者： A. Biswas A.C. den Brinker SPS Group (EH-03) Technische Universiteit Eindhoven Eindhoven Netherlands DSP Group (WO-02) Philips Natuurkundig Laboratorium Eindhoven Netherlands

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Effective high frequency regeneration based on sinusoidal modeling for MPEG-4 HE-AAC

Effective high frequency regeneration based on sinusoidal mo...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Sang-Uk Ryu K. Rose Joon-Hyuk Chang Electrical and Computer Engineering University of California Santa Barbara CA USA Korea Institute of Science and Technology Imaging Media Research Center Seoul South Korea

A novel approach is proposed for effective high frequency regeneration in audio coding, which is based on a sinusoids plus noise model. It assumes a standard high efficiency advanced audio coding (HE-AAC) encoder, and modifies the decoder to exploit all available information in estimating the model parameters. From the lower band reconstruction of core AAC, frequency parameters of the high band sinusoids are estimated. Side information about spectral energy and the regenerated high band of standard HE-AAC are employed in estimating the magnitude parameters of the high band sinusoids as well as noise model parameters. The gains achieved by the proposed technique, over conventional HE-AAC, are demonstrated by subjective quality tests that were carried out on audio signals with significant harmonics in the high band.

关键词： MPEG 4 Standard Frequency estimation Decoding audio coding Parameter estimation Image reconstruction Power harmonic filters Filter bank Testing Degradation

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