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检索条件"主题词=Audio coding"

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An iterated rational filter bank for audio coding

An iterated rational filter bank for audio coding

引用

IEEE-SP International Symposium on Time-Frequency and Time-Scale Analysis

作者： T. Blu France-Télécom-CNET Issy-les-Moulineaux France

This paper proposes a regular third-of-an-octave filter bank for high fidelity audio coding. The originality here is twofold: first, the filter bank is an iterated orthonormal rational filter bank for which the generating filters have been designed so that its outputs closely approximate a wavelet transform. This is different from the known coding algorithms which all use an integer filter bank, and most often a uniform one; second, the masking procedure itself is modelized with the help of a wavelet transform unlike the classical procedure in which a short time spectrum is computed and which gives rise to unwanted preecho effects. The masking procedure is then made equivalent to a quantization procedure. A simple non-optimized algorithm has been worked out in order to show the benefits of such a structure, especially in terms of preecho (which is perceptually inaudible), and the disadvantages, especially as far as delay is concerned.

关键词： Filter bank audio coding Discrete wavelet transforms Finite impulse response filter Wavelet transforms Algorithm design and analysis Ear Frequency Wavelet analysis Discrete transforms

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Individualization in spatial-audio coding

Individualization in spatial-audio coding

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： C. Jin N. van Schaik V. Best S. Carlile School of EIE University of Sydney Sydney NSW Australia Department of Physiology University of Sydney Sydney NSW Australia

A novel spatial-audio coding technique that allows individualized 3D audio presentation is described. It exploits the dichotomous roles of the low-frequency interaural timing and level difference cues versus the high-frequency spectral cues in human sound localization. The high-frequency spectral cues are modified to match the acoustics of the listener's outer ears, while preserving the original low-frequency interaural cues. The theory behind the coding technique is described and sound localization data are shown demonstrating the fidelity of the coding technique. Issues relating to the frequency resolution of directional frequency bands are explored.

关键词： Humans Ear Frequency audio coding Auditory system Australia Filters Acoustic measurements Transfer functions Timing

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On the Characteristics of Various Quranic Recitation for Lossless audio coding Application

On the Characteristics of Various Quranic Recitation for Los...

引用

International Conference on Computer and Communication Engineering, ICCCE

作者： Teddy Surya Gunawan Mira Kartiwi Electrical & Computer Engineering Department IIUM Kuala Lumpur Malaysia SEET UNSW Sydney Australia Information Systems Department International Islamic University Malaysia Kuala Lumpur Malaysia

ISBN: (纸本)9781509024285

Nowadays, there are more than 500 Quranic recitations available freely on the Internet. Most of the audio files were recorded in MP3 format due to its popularity, but lossless audio coding has gained much interest due to its high quality. The drawback of the lossless audio coding is the higher bitrate requirement. In this paper, the characteristics of the Quranic recitation of all surahs from various Reciters were evaluated. Results show that on average the voiced to unvoiced speech ratio is 21.39 for Quranic recitation, while the English audiobook on average has ratio of 7.49. It means that the voiced speech is more dominant compared to unvoiced speech in Quranic recitation. It is expected that the linear predictor component in a typical lossless audio compression could be optimized further so that efficient representation of Quranic audio signal could be achieved.

关键词： audio coding Computers Databases Hardware Software History Standards

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Warped linear predictive audio coding in video conferencing application

Warped linear predictive audio coding in video conferencing ...

引用

European Signal Processing Conference (EUSIPCO)

作者： Kalle Palomäki Akt Härmä Unto K. Laine Aalto-yliopisto Insinooritieteiden korkeakoulu Aalto FI

A codec for wideband 12kHz speech and audio in a video conferencing application is proposed in this paper. The codec is based on warped linear predictive coding algorithm which utilizes the auditory Bark frequency resolution. The structure of the codec is described and the main issues on the real-time implementation are discussed. The codec is integrated to a video conferencing product which is a video codec PC-board. The algorithm is implemented in a Texas TMSC31 digital signal processor.

关键词： Codecs Speech Speech coding Quantization (signal) Bandwidth audio coding

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Different Quantisation Noise Shaping Methods for Predictive audio coding

Different Quantisation Noise Shaping Methods for Predictive ...

引用

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

作者： S. Wabnik G. Schuller J. Hirschfeld U. Kraemer Fraunhofer Institute for Digital Media Technology Ilmenau Germany

The paper presents a comparison of a previous and a new approach to shape quantization noise in low bit rate predictive audio coding. The previous approach uses an adaptation of the step size of a uniform quantizer, the new approach uses a quantizer with clipping. Both approaches are evaluated using a predictive audio coding scheme. The presented results of a listening test show the improved performance of the new approach

关键词： Quantization Noise shaping audio coding Delay Acoustic noise Masking threshold Bit rate Frequency Decoding Testing

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An Efficient DSP Implantation of Wavelet audio coding for Digital Communication

An Efficient DSP Implantation of Wavelet Audio Coding for Di...

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International Conference on the Digital Society

作者： Gaoyong Luo School of Computing and Advanced Technologies Buckinghamshire New University High Wycombe Buckinghamshire UK School of Physics and Electronic Engineering Guangzhou University Guangzhou Guangdong China

ISBN: (纸本)9781424458059;9780769539539

Digital audio offers increased sound quality and greater signal processing flexibility than its analog counterpart. The key enabling technology for digital audio wireless products is audio compression due to channel bandwidth constraints. However, current audio coding schemes can hardly achieve ultra-low delay of encoding and decoding for live productions. This paper presents an efficient DSP implantation of wavelet audio coding using very short block processing to meet the very low delay requirement. The audio signal decomposition and reconstruction is performed by a two dimensional (2D) spatial-frequency processing of lifting wavelet transform to fully exploits the correlation for better compression performance. The lifting wavelet with boundary effects minimized is developed with lifting coefficient optimally quantized and implanted by fixed-point arithmetic applying only bit shifting and addition operations to replace multiplications and divisions, thus minimizing the computational complexity for real-time applications. A modified 2D embedded SPIHT algorithm with more bits used to encode the wavelet coefficients and transmitting fewer bits in the sorting pass, is implemented in fixed-point computation. Experimental results demonstrate that the proposed coder is efficient and has low complexity with less memory requirements for digital communication.

关键词： Digital signal processing audio coding Digital communication Acoustic signal processing audio compression Bandwidth Decoding Production Propagation delay Signal resolution

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A QUASI-ORTHOGONAL, INVERTIBLE, AND PERCEPTUALLY RELEVANT TIME-FREQUENCY TRANSFORM FOR audio coding

A QUASI-ORTHOGONAL, INVERTIBLE, AND PERCEPTUALLY RELEVANT TI...

引用

European Signal Processing Conference

作者： Olivier Derrien Thibaud Necciari Peter Balazs Universite de Toulon and Laboratoire de Mecanique et d'Acoustique - CNRS Acoustics Research Institute - OAW

ISBN: (纸本)9781479988518

We describe ERB-MDCT, an invertible real-valued time-frequency transform based on MDCT, which is widely used in audio coding (e.g. MP3 and AAC). ERB-MDCT was designed similarly to ERBLet, a recent invertible transform with a resolution evolving across frequency to match the perceptual ERB frequency scale, while the frequency scale in most invertible transforms (e.g. MDCT) is uniform. ERB-MDCT has mostly the same frequency scale as ERBLet, but the main improvement is that atoms are quasi-orthogonal, i.e. its redundancy is close to 1. Furthermore, the energy is more sparse in the time-frequency plane. Thus, it is more suitable for audio coding than ERBLet.

关键词： Non-stationary time-frequency transforms ERB filters MDCT audio coding

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High quality switched wavelet packet and discrete cosine transform audio coding technique

High quality switched wavelet packet and discrete cosine tra...

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IEEE International Symposium on Intelligent Signal Processing and Communication Systems (ISPACS)

作者： Pao-Chi Chang Tzung-Hung Wu Kong-Yueh Ho Department of Communication Engineering National Central University Zhongli Taiwan

We propose a hybrid coding system that utilizes both wavelet packet (WP) and DCT techniques. To process each audio frame, the system selects either WP or DCT as the transform based on the frame flatness measures in the wavelet domain and the frequency domain. If DCT is chosen, all DCT coefficients are quantized by a non-uniform quantizer according to the frequency masking curve. On the other hand, frame data are segmented into 26 fixed subbands when WP is chosen. Then, the system selectively utilizes DCT to promote frequency resolution of each subband based on the subband flatness measure. By quoting optimal bit-allocation for non-ideal filter bank, the masking threshold from psychoacoustic model can be translated into specific criteria in the wavelet domain for quantization. Experiment results show that the proposed system is superior to MP3 and AAC LC profile at 64k bps.

关键词： Packet switching Discrete wavelet transforms Wavelet packets Discrete cosine transforms audio coding Frequency measurement Wavelet domain Frequency domain analysis Filter bank Masking threshold

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High Quality audio coding with Mdctnet

High Quality Audio Coding with Mdctnet

引用

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

作者： Grant Davidson Mark Vinton Per Ekstrand Cong Zhou Lars Villemoes Lie Lu Dolby Laboratories Inc. San Francisco CA USA Dolby Sweden AB Stockholm Sweden

We propose a neural audio generative model, MDCTNet, operating in the perceptually weighted domain of an adaptive modified discrete cosine transform (MDCT). The architecture of the model captures correlations in both time and frequency directions with recurrent layers (RNNs). An audio coding system is obtained by training MDCTNet on a diverse set of fullband monophonic audio signals at 48 kHz sampling, conditioned by a perceptual audio encoder. In a subjective listening test with ten excerpts chosen to be balanced across content types, yet critical for both codecs, the mean performance of the proposed system for 24 kb/s variable bitrate (VBR) is similar to that of Opus at twice the bitrate.

关键词： Training Adaptation models Time-frequency analysis Codecs audio coding Bit rate Signal processing

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Soft decision unequal error protection scheme for MPEG advanced audio coding

Soft decision unequal error protection scheme for MPEG advan...

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IEEE International Conference on Multimedia and Expo (ICME)

作者： T.H. Yeo W.C. Wong D.Y. Huang Department of Electrical Engineering National University of Singapore Singapore Institute for Infocomm Research National University of Singapore Singapore

In this paper, a system using soft decision decoding of convolutional codes is investigated for MPEG-4 advanced audio coding. Under severe channel random bit error rates of 4.00x10/sup -2/ and above, the proposed scheme is able to achieve above 90% improvement in residual bit error rate performance over the current scheme. The average percentage overhead incurred using the proposed scheme is about 21%, which is the same as that of the original UEP scheme. An analysis with the theoretical bounds shows that within experimental error, the proposed scheme can perform to near optimum as compared to the current scheme.

关键词： Error correction codes audio coding Codecs Decoding Convolutional codes MPEG 4 Standard Bit error rate Robustness Redundancy Forward error correction

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