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检索条件"主题词=Audio coding"

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Scalable superwideband extension for wideband coding

Scalable superwideband extension for wideband coding

引用

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

作者： Mikko Tammi Lasse Laaksonen Anssi Ramo Henri Toukomaa Nokia Research Center Tampere Finland

ISBN: (纸本)9781424423538

Recent trends in speech and audio codec standardization include scalability and extending the signal bandwidth beyond wideband (WB) to superwideband (SWB). In this paper we introduce a SWB extension for the ITU-T G.718 WB codec. In the SWB extension the high frequency content is generated utilizing the quantized MDCT domain coefficients of the WB core, which enables low additional delay. The proposed implementation is scalable with 4 kbps layers. In the first layer two different coding modes are used depending on the input signal type. The proposed SWB extension is evaluated with listening tests and complexity analysis.

关键词： Wideband Frequency Bandwidth Scalability Speech coding Streaming media Delay Speech codecs audio coding Standardization

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Hardware accelerating audio coding algorithms

Hardware accelerating audio coding algorithms

引用

IEEE International Conference on Consumer Electronics (ICCE)

作者： F. Mayer D. Dalquen T. Dettbarn Fraunhofer Institute of Integrated Circuits and Systems Erlangen Germany

This paper evaluates hybrid hardware-software structures for acceleration and power savings in audio coding algorithms. Using the well known MPEG I/II layer 3 (mp3) decoder as example, we demonstrate the general principles and trade-offs. The results, obtained for mp3, are valid for (but not limited to) the full family of MPEG codecs, due to algorithmic similarities, including AAC and HE-AAC

关键词： Hardware Acceleration audio coding Tellurium Decoding Codecs Microwave integrated circuits Digital communication Art Costs

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Embedded transform coding of audio signals by model-based bit plane coding

Embedded transform coding of audio signals by model-based bi...

引用

33rd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Hoang, Thi Minh Nguyet Oger, Marie Ragot, Stephane Antonini, Marc France Telecom R&D TECH SSTP Av Pierre Marzin F-22307 Lannion France Univ Nice Sophia Antipolis F-06903 Sophia Antipolis France CNRS UMR 6070 Lab I3S F-06903 Sophia Antipolis France

ISBN: (纸本)9781424414833

This paper proposes a new model-based method for transform coding of audio signals. The input signal is mapped in "perceptual" domain by linear-predictive weighting filter followed by modified discrete cosine transform (MDCT). To provide bitstream scalability, model-based bit plane coding is then applied with respect to the mean square error (MSE) criterion. We present methods to estimate the symbol probability in bit planes assuming a generalized Gaussian model for the distribution of MDCT coefficients. We compare the performance of the proposed bitstream scalable coder with stack-run coding and ITU-T G.722.1. Objective and subjective quality results are presented. The proposed coder is equivalent to or slightly worse than reference coders, but presents the nice advantage of being scalable. Performance penalty due to bitstream scalability is evident at low bitrates.

关键词： transform coding audio coding

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Multiple description spherical quantization of sinusoidal parameters with repetition coding of the amplitudes

Multiple description spherical quantization of sinusoidal pa...

引用

Asilomar Conference on Signals, Systems & Computers

作者： Jesper Rindom Jensen Mads Græsbøll Christensen Morten Holm Jensen Søren Holdt Jensen Torben Larsen Department of Electronic System University of Aalborg Denmark Department of Media Technology University of Aalborg Denmark Widex A/S Vaerlose Denmark

ISBN: (纸本)9781424458257

Recently, multiple description spherical trellis-coded quantization (MDSTCQ) for quantization of sinusoidal parameters was proposed, which suffered from a suboptimal implementation. Therefore, we propose the multiple description spherical quantization with repetition coding of the amplitudes (MDSQRA) scheme. Here, we apply repetition coding to the amplitudes, whereas the phases and the frequencies are coded using multiple description quantization. We measure the relation between the expected perceptual distortions for the two quantization schemes, which shows that MDSQRA outperforms MDSTCQ for most packet-loss probabilities. Also, we measure the perceptual distortion obtained when applying MDSQRA on synthetic data and show that it matches the expected perceptual distortion. Furthermore, due to MDSTCQs suboptimal implementation, MDSQRA outperforms MDSTCQ in terms of measured perceptual distortion even at packet-loss probabilities where the expected perceptual distortion was lowest for MDSTCQ.

关键词： Quantization audio coding Distortion measurement Frequency Robustness Decoding Masking threshold Redundancy Humans Ear

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Relationship between consecutive frames in generalized harmonics analysis for predictive coding

Relationship between consecutive frames in generalized harmo...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Hisayori Noda Akinori Nishihara Communications and Integrated Systems Tokyo Institute of Technology Japan The Center of Research and DeveLopment of Educational technology Tokyo Institute of Technology Japan

ISBN: (纸本)9781424438273

Relationship between consecutive frame in the generalized harmonics analysis and possibility of predictive coding is described. In the GHA, each frame of a signal is shown as sum of sinusoids. The parameters which show the same sinusoid are often extracted from two consecutive frames. We propose a method to extract the parameters which show the same sinusoid using cross correlation function and greedily selection algorithm. This algorithm is applied to several audio signals and it is confirmed that sinusoids are extracted across multiple consecutive frames. Possibility to apply the proposed method to predictive coding is also shown.

关键词： Harmonic analysis Predictive coding Frequency estimation Educational technology audio coding audio compression Fourier transforms Signal resolution Signal synthesis Parameter estimation

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On the pre-echo control method in transient signal coding of AVS audio

On the pre-echo control method in transient signal coding of...

引用

International Conference on audio, Language and Image Processing

作者： Zhang, Tao Wang, Wei He, Jialin Tianjin Univ Sch Elect & Informat Engn Tianjin Peoples R China

ISBN: (纸本)9781424417230

The audio and Video coding Standard of China(AVS) uses Multi-Resolution Analysis of Frequency Domain (MRAFD) to control pre-echo when coding transient signal. However this technology complicates the implementation of encoder as well as decoders with more operation and higher computational precision. In this paper a kind of gain control of time domain is used to encode transient signal instead of MRAFD. Compared with MRAFD, the new method greatly decreases the computational complexity of AVS audio encoder and decoder while the coding quality. of the two methods is almost the same.

关键词： Transient analysis Gain control Encoding Time domain analysis audio coding Bit rate Decoding

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An error robust Ultra Low Delay audio coder using an MA prediction model

An error robust Ultra Low Delay audio coder using an MA pred...

引用

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

作者： Stefan Wabnik Gerald Schuller Ferenc Kraemer Institute of Digital Media Technology Fraunhofer IDMT Ilmenau Germany Institute of Media Technology Technical University Ilmenau Ilmenau Germany

ISBN: (纸本)9781424423538

This paper compares two prediction structures for predictive perceptual audio coding in the context of the ultra low delay (ULD) coding scheme. One structure is based on the commonly used AR signal model, leading to an IIR predictor in the decoder. The other structure is based on an MA signal model, leading to an FIR predictor in the decoder. We find that the AR-based predictor has a slightly better performance in case of an undisturbed transmission channel, but the MA-based predictor has a much better performance in case of transmission errors. For a bit error rate (BER) of 1.0e-5, the perceptual quality of the proposed MA model predictor achieves a mean objective difference grade (ODG) of -0.66 ODG whereas the AR model predictor only reaches -3.42 ODG.

关键词： Robustness Delay Predictive models Decoding audio coding Predictive coding IIR filters Frequency Finite impulse response filter Bit error rate

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Design and Implementation of MPEG-4 AAC Decoder on ARM Embedded System for CMMB Receiver

Design and Implementation of MPEG-4 AAC Decoder on ARM Embed...

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International Conference on Management and Service Science, MASS

作者： Ming Yan Guorong Hu Jia Wang GxSOC Research Institute Communication University of China Beijing China Radio & TV Studio Centre Design Department Radio Film and Television Design and Research Institute Beijing China

CMMB (China mobile multimedia broadcasting) adopts MPEG-4 HE AAC (high efficiency advanced audio coding) as its sound coding technology in video broadcast. In this paper, we first introduce the decoding flow of MPEG-4 HE AAC and its two key technologies, SBR (spectral band replication) and interbank. The IMDCT function was optimized especially because of its complicated calculation. Then we implemented the decoder into ARM embedded system and analyzed the complexity of each function in Linux OS. Test results indicate that the decoder can decode HE AAC bitstream with real time on the embedded system with ARM920T core.

关键词： MPEG 4 Standard Decoding Embedded system Helium Multimedia communication Broadcast technology Digital multimedia broadcasting audio coding Linux System testing

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Simulation and Application of FFT Based Pipelined Stream

Simulation and Application of FFT Based Pipelined Stream

引用

International Conference on Information Engineering and Computer Science (ICIECS)

作者： Zhi Dong Yimeng Zhang Zhiping Huang Guilin Tang Chunwu Liu Nat. Univ. of Defence Technol. Changsha China

In order to reduce the calculation time of FFT, this paper analyzes the theory of FFT, and presents a new method to compute Cooley-Turkey FFT which is based on pipelined stream. After Matlab, ISE simulation and FPGA experiment, the results show that: this method uses only 42 us to compute 4096 points FFT which is less than 2/3 time comparing to previous methods. It offers a better DFT method for high speed digital signal processing.

关键词： Field programmable gate arrays Discrete Fourier transforms Digital signal processing Equations Analytical models Computational modeling Pipelines Signal processing algorithms Image processing audio coding

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A Novel Low Bit Rate audio Bandwidth Extension Method

A Novel Low Bit Rate Audio Bandwidth Extension Method

引用

International Symposium on Knowledge Acquisition and Modeling, KAM

作者： Bo Hang Ruimin Hu Ye Ma National Engineering Research Center of Multimedia Software Wuhan University of China Wuhan China Mathematics and Computer Science College Xiangtan University Xiangfan China

In present communication system, high quality audio signal is supposed to be provided with low bit rate and low computational complexity. This paper proposed a novel audio coding bandwidth extension method, which can improve decoded audio quality with increasing only a few coding bits per frame and a little computational complexity. This method calculate high-frequency synthesis filter by using codebook mapping method, and transmit only quantified gain corrections in high-frequency part of multiplexing coding bit stream. The preliminary test show that this method can provide comparable audio quality with lower bit consumption and computational complexity compared to the high frequency regeneration of AMR-WB+.

关键词： Bit rate Bandwidth Computational complexity audio coding Decoding Signal synthesis Filters Streaming media Testing Frequency

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