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Combined speech and audio coding by discrimination

Combined speech and audio coding by discrimination

引用

7th IEEE Workshop on Speech coding

作者： Tancerel, L Ragot, S Ruoppila, VT Lefebvre, R Univ Sherbrooke Dept Elect Engn Sherbrooke PQ J1K 2R1 Canada

ISBN: (纸本)0780364163

We propose in this paper a general solution for combined speech and audio coding. Particularly, we describe a speech/music discrimination procedure for multi-mode wideband coding. The speech/music decision is updated only when a low-energy frame is detected, and kept unchanged otherwise. The signal is classified using second-order statistics of discriminant parameters. An experimental CELP/transform coder operating at 16 kbit/s is demonstrated. Results show improved performance when compared to single-mode encoding.

关键词： audio coding Bit rate Delay estimation Forward contracts Linear predictive coding Multiple signal classification Speech coding Statistics Switches Wideband

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A QUASI-ORTHOGONAL, INVERTIBLE, AND PERCEPTUALLY RELEVANT TIME-FREQUENCY TRANSFORM FOR audio coding 23

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

引用

23rd European Signal Processing Conference (EUSIPCO)

作者： Derrien, Olivier Necciari, Thibaud Balazs, Peter Univ Toulon & Var Marseille France Lab Mecan & Acoust CNRS Marseille France Acoust Res Inst OAW Vienna Austria

ISBN: (纸本)9780992862633

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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FLEXCODE - FLEXIBLE audio coding

FLEXCODE - FLEXIBLE AUDIO CODING

引用

2010 IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Klejsa, Janusz Li, Minyue Kleijn, W. Bastiaan KTH Royal Inst Technol ACCESS Linnaeus Ctr S-10044 Stockholm Sweden

ISBN: (纸本)9781424442966

Modern networks are highly variable and, as a result, source coders are commonly used under conditions that they were not designed for. We address this problem with a source-coding philosophy that aims at the instantaneous re-optimization of a source coder to match a wide range of constraints on rate or quality and a wide range of packet-loss rates. We present a number of technologies that can be reconfigured by solving analytic relations that use the current conditions and a statistical description of the source as input. The technologies include distribution-preserving quantizers, flexible multiple-description quantizers, and a rate distribution scheme. Based on the generic technologies, we created a complete audio coder. Formal listening tests show that the resulting audio coding scheme with full flexibility provides a quality that is on-par with the best standardized codecs for any particular rate.

关键词： audio coding source modeling quantization

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SPATIAL-TEMPORAL MULTI-CHANNEL audio coding

SPATIAL-TEMPORAL MULTI-CHANNEL AUDIO CODING

引用

IEEE Sensor Array and Multichannel Signal Processing Workshop

作者： Lee, Jonghwa Lee, Chulhee Yonsei Univ Dept Elect & Elect Engn Seoul 120749 South Korea

ISBN: (纸本)9781424422401

This paper presents a multi-channel audio coding method compatible with the standard stereo audio coding method. The encoding method consists of three parts: spatial-temporal analysis of sound source location, compressing multi-channel audio into stereo audio and encoding stereo signals. The proposed method is simpler and more effective than conventional methods in terms of audio data size with small additional information. With proposed method, it is possible to transmit or store multi-channel audio signals as stereo audio signals.

关键词： Encoding ISO audio coding Position measurement Standards Transform coding Digital audio broadcasting

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Efficient implementation of a class of MDCT/IMDCT filterbanks for speech and audio coding applications

Efficient implementation of a class of MDCT/IMDCT filterbank...

引用

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

作者： Chivukula, Ravi K. Reznik, Yuriy A. QUALCOMM Inc San Diego CA 92121 USA Univ Texas Arlington Elect Engn Dept Arlington TX 76019 USA

ISBN: (纸本)9781424414833

In this paper, an efficient algorithm for implementing MDCT/IMDCT of lengths N = 5.2(m) (m >= 2) is presented. Transforms for such lengths are of interest for speech and audio coding applications, such as recently issued and/or emerging standards G.729.1, ***-VBR, and EVRC-WB. In our design we utilize a mapping of MDCT of size N into N/2-point DCT-IV and DCT-II with isolated pre-multiplications, which are subsequently moved in the windowing stage. We show that such a modified window is piece-wise symmetric, and can be stored using N/2 words. In our algorithm we also use an efficient factorization of 5-point DCT-II which requires only 4 multiplications by irrational factors. We compare our proposed algorithm with several alternative implementations and show that our design offers practically appreciable reduction in complexity and memory usage.

关键词： speech coding audio coding FFT DCT MDCT factorization fast algorithms

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Burrows-Wheeler Transform and Combination of Move-to-Front coding and Run Length Encoding for Lossless audio coding 9

Burrows-Wheeler Transform and Combination of Move-to-Front C...

引用

9th International Conference on Computer Engineering & Systems (ICCES)

作者： Elsayed, Hend. A. Delta Univ Sci & Technol Fac Engn Dept Commun & Comp Engn Mansoura Egypt

ISBN: (纸本)9781479965946

This paper presents a lossless audio coding using Burrows-Wheeler Transform (BWT) and a combination of a Move-To-Front coding (MTF) and Run Length Encoding (RLE). audio signals used are assumed to be of floating point values. The BWT is applied to this floating point values to get the transformed coefficients;and then these resulting coefficients are converted using the Move-to-Front coding to coefficients can be better compressed and then these resulting coefficients are compressed using a combination of the Run Length Encoding, and entropy coding. Two entropy coding are used which are Arithmetic and Huffman coding. Simulation results show that the proposed lossless audio coding method outperforms other lossless audio coding methods;using only Burrows-Wheeler Transform method, using combined Burrows-Wheeler Transform and Move-to-Front coding method, and using combined Burrows-Wheeler Transform and Run Length Encoding method.

关键词： audio coding Burrows-Wheeler Transform Move-To-Front coding Run Length Encoding Entropy coding Arithmetic coding Huffman coding

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GAUSSIAN-MIXTURE MODELING OF LATTICE-BASED SPHERICAL VECTOR QUANTIZATION PERFORMANCE IN TRANSFORM audio coding

GAUSSIAN-MIXTURE MODELING OF LATTICE-BASED SPHERICAL VECTOR ...

引用

2010 IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Patchoo, Wisarn Fischer, Thomas R. Washington State Univ Sch EECS Pullman WA 99163 USA

ISBN: (纸本)9781424442966

A block-based Gaussian mixture model (GMM) is used to model the distribution of transform audio data to be encoded using lattice-based spherical vector quantization (LSVQ). The expectation-maximization algorithm is used to design the GMM to model the marginal density of the transform coefficients and the vector energy density. A GMM-based rate-distortion function is derived and shown to closely match the observed spherical VQ performance. The LSVQ transform audio coding performance is characterized for the best lattices known in 4, 8, 16, and 32 dimensions.

关键词： Gaussian distributions audio coding rate distortion theory and vector quantization

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IMPROVED ARITHMETIC coding FOR TIME-WARPED MDCT BASED audio coding

IMPROVED ARITHMETIC CODING FOR TIME-WARPED MDCT BASED AUDIO ...

引用

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

作者： Bayer, Stefan Edler, Bernd Int Audio Labs Erlangen D-91058 Erlangen Germany

ISBN: (纸本)9781479903566

The Time-Warped Modified Discrete Cosine Transform (TW-MDCT) improves the energy compaction for harmonic signals with varying fundamental frequency compared to the plain MDCT. Adaptive context based entropy coding has the potential to provide higher gain over memoryless entropy coding. But in combination with the TW-MDCT, the context based adaptive coding may lead to suboptimal coding. This paper presents an algorithm for improving the context for the TW-MDCT. This is mainly achieved by exploiting already available information on the frequency variation needed by the TW-MDCT. This results in an improved entropy coding.

关键词： audio coding Entropy coding TW-MDCT Arithmetic coding

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The Embedded audio coding of Algebraic Quantization Based on the Transform Coefficients

The Embedded Audio Coding of Algebraic Quantization Based on...

引用

2nd International Conference on Mechanical Engineering and Green Manufacturing (MEGM 2012)

作者： Tang, Gang Huangshi Inst Technol Huangshi 435003 Peoples R China

ISBN: (纸本)9783037853771

This paper proposes a new quantitative method called the transform coefficients, which is based on the algebraic quantization method. This coefficient is defined as a unique amplitude multiplied by a number of *** order to determine the signs of pulse, location and amplitude, we selected the transmission coefficient by adopting an ideal standard errors. This simple quantification has been implemented in scalable coding of microwave transform, and has been proved that its voice signal and music effect is better than that of the SPIHT.

关键词： Embedded coding audio coding Algebraic quantization

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Optimal bit reservoir control for audio coding

Optimal bit reservoir control for audio coding

引用

Workshop on Applications of Signal Processing to audio and Acoustics

作者： Camberlein, E Philippe, P France Telecom R&D F-35512 Cesson Sevigne France

ISBN: (纸本)0780391543

audio coding schemes such as MP3 or MPEG AAC can use a variable number of bits to encode each signal frame. However they can still be used oil constant bit rate channels thanks to a variation buffer limited in size. This buffer allows the encoder to use more bits for high complexity frames in order to maintain quality, the buffer being decreased e.g. for silent frames, leading to a constant rate coding on average. The use of such a buffer aims at reducing quality fluctuations over time, thanks to reasonable instantaneous bit rate variations. In this paper, a new design for a buffer controller is presented. This controller aims at finding the appropriate number of bits to encode each audio frame, according to chosen perceptual criteria and reduces tile perceived distortion over time. This process can take place in a two pass encoding process, the First pass aiming at measuring frames complexity and bit demand, the second consisting in allocating the appropriate number of bits for each. In order to cope with online coding application a single pass encoding with a reasonable additional delay is also investigated.

关键词： audio coding optimal control MP3 MPEG buffer controller constant bit rate channels online coding application optimal bit reservoir control quality fluctuation reduction audio coding Digital audio players Moving Pictures Experts Group Optimal control coding Drill bits

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