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检索条件"主题词=Audio coding"

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Perceptually Controlled Error Protection for audio Streaming over IP Networks

Perceptually Controlled Error Protection for Audio Streaming...

引用

International Conference on Digital Telecommunications (ICDT)

作者： E. Hellerud J.E. Voldhaug U.P. Svensson Centre for Quantifiable Quality of Service in Communication Systems Norwegian University of Science and Technology Trondheim Norway

This paper presents an unequal error protection (UEP) scheme for audio streaming over IP networks. The sender simulates packet loss by employing the same error concealment scheme as the receiver for each audio frame. The perceived quality of such an estimated frame is compared to that of a lower bit rate version. Whenever the quality of the estimated frame is below a given threshold relative to a lower bit rate version, the frame is error protected by sending the low bit rate version in addition. Simulations have been performed using MPEG-4 advanced audio coding (AAC) encoding, perceptual evaluation of audio quality (PEAQ) as quality measurement tool, and the gapped data amplitude and phase estimation (GAPES) algorithm as error concealment. Results show that by adjusting the threshold level for when frames are error protected, the quality improvement vs. redundancy can be quantified

关键词： Error correction Protection Streaming media IP networks Bit rate Error correction codes Performance evaluation MPEG 4 Standard audio coding Encoding

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Quantization of transmission parameters in stereo linear predictive systems

Quantization of transmission parameters in stereo linear pre...

引用

Data Compression Conference (DCC)

作者： A. Biswas A.C. den Brinker Technical University Eindhoven Eindhoven Netherlands Philips Research Laboratories Eindhoven Netherlands

Linear prediction (LP) is a widely used technique in single-channel speech coding. LP can also be applied to stereo (or multi-channel) audio coding, but it requires a scheme for quantization and efficient transmission of stereo-LP parameters. We propose to transmit the (forward) normalized reflection matrices together with the zero-lag correlation matrix. Furthermore, we select a parameterization for the matrices and a quantization strategy per parameter. The parameterization for the normalized reflection matrices is a variant of the singular value decomposition (SVD). The proposed quantization scheme is evaluated for first-order stereo-LP systems. As criterion, we use a spectral distortion (SD) measure based on the norm of the transfer matrix of the synthesis filter. Simulations show that the distortions can easily be controlled as a function of bit rate

关键词： Quantization Matrix decomposition Reflection Speech coding audio coding Singular value decomposition Distortion measurement Control system synthesis Filters Bit rate

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Encoder Assisted Frame Loss Concealment for MPEG-AAC Decoder

Encoder Assisted Frame Loss Concealment for MPEG-AAC Decoder

引用

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

作者： Sang-Uk Ryu E. Choy K. Rose ECE Department University of California Santa Barbara CA USA Qualcomm Inc. San Diego CA USA

This paper proposes an efficient method for frame loss concealment within the advanced audio coding (AAC) decoder, which can effectively mitigate the adverse impact of transmission errors on the reconstruction quality. The lost frame information is estimated in the modified discrete cosine transform (MDCT) domain in terms of magnitude and sign of the coefficients. A computationally efficient approach that is capable of providing accurate estimation is employed for the magnitude estimate. Enhanced sign estimation is developed and implemented, which exploits extra information from encoder. Extensive subjective quality evaluation demonstrates that the proposed method achieves substantial quality improvement with minimal encoder assistance

关键词： Decoding audio coding Discrete cosine transforms Error correction Costs Propagation losses Laboratories Acoustic noise Degradation Interpolation

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Low Complex audio Encoding for Mobile, Multimedia

Low Complex Audio Encoding for Mobile, Multimedia

引用

IEEE Conference on Vehicular Technology (VTC)

作者： J. Makinen A. Lakaniemi P. Ojala Multimedia Technologies Laboratory Nokia Research Center Finland

Computational resources in a mobile terminal and in wireless transmission channels set strict requirements for audio coding in mobile multimedia applications. This paper introduces, a low complex audio encoding method enabling high-quality audio for mobile multimedia. The presented method is the low complexity version of the extended AMR-WB audio encoder in the 3GPP standards. The encoder was designed based on requirements of the 3GPP multimedia messaging service (MMS)

关键词： Encoding Bit rate Frequency audio coding Speech codecs Streaming media Sampling methods Bandwidth Signal processing Message service

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STREAMING OVER THE INTERNET WITH A SCALABLE PARAMETRIC audio CODER

STREAMING OVER THE INTERNET WITH A SCALABLE PARAMETRIC AUDIO...

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European Signal Processing Conference

作者： J.C. Cuevas-Martinez P. Vera-Candeas N. Ruiz-Reyes University of Pisa

audio compression has progressively gained higher importance in the Internet thanks to massive amount of multimedia services on it. This new services require coders adapted to that new environment. Therefore, new generation coders use more complex models focused on features which make possible its use for audio streaming over the Internet, mainly low bit rate, scalability and robustness. In our case, a good trade-off between bit rate reduction and audio quality is achieved by using parametric audio coding, and furthermore, this coder has a scalable version, optimized for streaming requirements. This coder avoids differential information between coded audio segments and uses a layered scheme for changing straightforwardly the bit rate. The results reveal our coder as a good candidate for massive distributed audio applications, like music on demand, radio broadcasting or real-time streaming audio. In this article are shown the main features of this coder and their implication on streaming.

关键词： Internets audio audio coding audio streaming Bit rate bit rate reduction Multimedia services Music

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Expert system for intelligent audio codification based in speech/music discrimination

Expert system for intelligent audio codification based in sp...

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International Workshop on Genetic and Evolving Systems, GEFS

作者： J. E. Munoz Exposito S. Garcia Galan N. Ruiz Reyes P. Vera Candeas F. Rivas Pena Electronics and Telecommunication Engineering Department University of Jaén Linares (Jaén) Spain

Automatic speech/music discrimination has become a research topic of interest in the last years. This paper presents a new approach for speech/music discrimination, which is based on an expert system that incorporates fuzzy rules into its knowledge base. The proposed scheme consists of three stages: 1) features extraction, 2) audio signal classification, and 3) selection of the best audio coder every 23 ms. The fuzzy expert system improves the accuracy rate of a GMM classifier when included into the classification stage. In order to select the best audio coder, the expert system takes information of the current and past frames into account. It is important to emphasize that the low computational cost of the proposed approach makes it feasible for real time applications

关键词： Expert systems Intelligent systems Feature extraction Speech analysis Multiple signal classification audio coding Fuzzy systems Hybrid intelligent systems Pattern classification Speech coding

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Efficient Quantization Of Statistically Normalized Vectors Using Multi-Option Partial-Order Bit-Assignment Schemes

Efficient Quantization Of Statistically Normalized Vectors U...

引用

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

作者： S.A. Ramprashad DoCoMo Communications Laboratory USA Inc. San Jose CA USA

In this paper we focus on new options for the efficient quantization of statistically normalized target vectors at low bitrates. This problem is fundamental to many low-rate speech and audio coder designs. Here many such coders follow a general principle of taking a structured speech or audio signal, applying a process of redundancy removal and then quantizing each of the resulting statistically normalized targets to a relevant distortion level. We look at this latter problem when some of these targets are to be quantized at very low bitrates ( 1 bit/target-scalar). The approach we take is to efficiently communicate a target-adaptive pattern of unequal bit-assignments (noise allocations) across each target. This can increase performance over an approach that has a constant noise allocation even when target vectors consist of independent and identically distributed (i.i.d.) scalars. We extend these schemes to multi-option schemes allowing further options to adapt and improve performance

关键词： Quantization Bit rate Redundancy Speech processing Signal processing Decoding Distortion Signal design Speech coding audio coding

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A prototype system for object coding of musical audio

A prototype system for object coding of musical audio

引用

Workshop on Applications of Signal Processing to audio and Acoustics

作者： Vincent, E Plumbley, MD Queen Mary Univ London Dept Elect Engn London E1 4NS England

ISBN: (纸本)0780391543

This article deals with low bitrate object coding of Musical audio, and more precisely with the extraction of pitched sound objects in polyphonic music. After a brief review of existing methods, we discuss the potential benefits of recasting this problem in a Bayesian framework. We define pitched objects by a set of probabilistic priors and derive efficient algorithms to infer active objects and their parameters. Preliminary experiments suggest that the proposed method results in it better sound quality than simple sinusoidal coding while achieving a lower bitrate.

关键词： Bayes methods audio coding music Bayesian framework musical audio object coding polyphonic music sinusoidal coding sound quality

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Rate-distortion optimal high-resolution differential quantisation for sinusoidal coding of audio and speech

Rate-distortion optimal high-resolution differential quantis...

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Workshop on Applications of Signal Processing to audio and Acoustics

作者： Heusdens, R Jensen, J Korten, P Vafin, R Delft Univ Technol Dept Mediamat NL-2628 AL Delft Netherlands

ISBN: (纸本)0780391543

Sinusoidal coding plays an important role in low-rate audio coding. Typically, (time/frequency) differential techniques are employed to reduce the bit ride for representing the sinusoidal components. In this paper we derive optimal entropy-constrained differential (quantisers for quantising the sinusoid parameters. More specifically, the quantisers minimise a perceptually relevant distortion measure while the corresponding quantisation indices satisfy an entropy constraint. The quantisers turn Out to be flexible and of low complexity. Subjective evaluations with audio signals suggest a bit-rate reduction as high as 20% with the derived quantisers over state-of-the-art (logarithmic) quantisers.

关键词： audio coding rate distortion theory speech coding bit rate reduction bit-rate reduction rate-distortion optimal high-resolution differential quantisation sinusoid parameter quantisation sinusoidal coding time-frequency differen

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MDCT IP Core Generator with Architectural Model Simulation

MDCT IP Core Generator with Architectural Model Simulation

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IFIP International Conference on Very Large Scale Integration (VLSI-SoC)

作者： Peter Malik Marcel Balaz Tomas Pikula Martin Simlastik Institute of Informatic Slovak Academy of Sciences Bratislava Slovakia Institute of Informatics Slovak Academy of Sciences Bratislava Slovakia

Compression of digital audio signals has become increasingly more important with the advent of fast and inexpensive microprocessors and digital signal processors. Several compression schemes were developed and well established. Most of them adopt MDCT/IMDCT. This paper presents a new MDCT IP core generator. The software tool generates several MDCT architectures with adjustable parameters for FPGA-based design as well as computation precision and area estimations

关键词： Digital audio players Signal processing audio coding Circuits Digital signal processors Software tools Computer architecture audio compression Signal processing algorithms Frequency

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