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检索条件"主题词=Audio coding"

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Distributed Spatial audio coding in Wireless Hearing Aids

Distributed Spatial Audio Coding in Wireless Hearing Aids

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Olivier Roy Martin Vetterli School of Computer and Communication Sciences Ecole Polytechnique Fédérale de Lausanne Lausanne Switzerland Department of Electrical Engineering and Computer Sciences University of California Berkeley Berkeley CA USA

The information content of binaural signals can be beneficial to many algorithms deployed in current digital hearing aids. However, the exchange of such signals over a wireless communication link requires transmission schemes that must fulfill demanding technical constraints. We present a distributed coding algorithm that builds on psychoacoustic principles in order to achieve this goal with low bitrates, while preserving affordable complexity. The key steps of the proposed algorithm are detailed and the accuracy of the signal exchange mechanism is evaluated in simple simulated acoustic scenarios.

关键词： audio coding Hearing aids Auditory system Signal processing algorithms Bit rate Acoustic devices Distributed computing Wireless communication Psychology Psychoacoustic models

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ENHANCED LONG-TERM PREDICTOR FOR UNIFIED SPEECH AND audio coding

ENHANCED LONG-TERM PREDICTOR FOR UNIFIED SPEECH AND AUDIO CO...

引用

IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Jeongook Song Hyen-O Oh Hong-Goo Kang Digital Signal Processing Lab Yonsei University Seoul Korea

Unified Speech and audio coding (USAC) is an emerging MPEG audio standard striving for efficiently representing both speech and music signals even in very low bitrate ranges. The reference codec takes an approach of unifying two state-of-the-art speech and audio coding structures in a single platform. This paper proposes an enhanced long term predictor (eLTP) that effectively utilizes periodic redundancies of inter- and intra- time frames. Experimental results with various types of input signals confirm the superiority of the proposed algorithm compared to the reference codec.

关键词： Speech Codecs Transform coding audio coding Speech coding Decoding

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MDCT domain parametric stereo audio coding

MDCT domain parametric stereo audio coding

引用

International Conference on Signal Processing and Communications, SPCOM

作者： K. Suresh R Akhil Raj Dept. of Electronics & Communication Engg. College of Engineering Thiruvananthapuram India Dept. of Electronics & Communication Engg. College of Engineering Thiruvananthapuram 695016 India

Parametric coding of multichannel audio has gained popularity for low bit rate audio coding applications such as digital audio broadcasting. Most of the existing algorithms use MDCT domain techniques for compressing the audio, while the spatialization parameter estimation is done in a different time-frequency domain. An MDCT domain parametric stereo coding algorithm which represents the stereo channels as the linear combination of the `sum' channel derived from the stereo channels and a reverberated channel generated from the `sum' channel has been reported in literature. Spatialization parameters are estimated at the encoder by taking the scaled sub-band projections of stereo channels on `sum' and reverberated channel. This model is inadequate to represent the stereo image since only four parameters per sub-band are used as spatialization parameters. In this work we improve the perceptual quality of this MDCT domain parametric coder with an augmented parameter extraction scheme using an additional reverberated channel. Subjective evaluation using MUSHRA test illustrates that the new algorithm has increased the perceptual audio quality of the encoded audio signal significantly.

关键词： Delay audio coding Parameter extraction Decoding Approximation methods Channel estimation

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Multi-sinusoid excitation model for audio coding

Multi-sinusoid excitation model for audio coding

引用

International Conference on Communication Systems

作者： Wen-Whei Chang Li-Wei Wang Department of Communication Engineering National Chiao Tung University Hsinchu Taiwan

ISBN: (纸本)0780320468

Most studies in LPC-based audio coders decompose the signal into the product of excitation and system spectra, and then quantize the excitation by using either a stochastic codebook or multiple pulses. But their near white spectra cannot precisely describe the harmonic characteristics of excitation, especially when dealing with instrumental music. The paper explores the benefits of sinusoidal representation for excitation in the design of analysis-by-synthesis predictive coders. Furthermore, an efficient parameter extraction algorithm has also been developed to identify the associated parameters of the sinusoidal components. Simulation results indicate that the proposed multi-sinusoid excitation model allows the implementation of an LPC-based audio coder which delivers near toll quality at the rate of 92.61 kbps.< >

关键词： audio coding Linear predictive coding Psychoacoustic models Frequency estimation Predictive models Stochastic systems Power harmonic filters Instruments Citation analysis Transform coding

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Rate-distortion allocation for time-frequency dependent audio coding

Rate-distortion allocation for time-frequency dependent audi...

引用

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

作者： R. Der P. Kabal Wai-Yip Chan Electrical & Computer Engineering McGill University Montreal QUE Canada Electrical & Computer Engineering Queen's University Kingston ONT Canada

A stream coding framework is presented for solving the distortion-constrained time-frequency dependent quantization problem that naturally arises when overlapped time-frequency decompositions are used. The main contributions of this paper are: (1) an efficient rate-distortion allocation algorithm for dependent quantization when the neighborhood of dependency is large; and (2) demonstration that a perceptual excitation distortion measure produces better coded audio quality than the conventional noise-to-mask ratio measure.

关键词： Rate-distortion Time frequency analysis audio coding Quantization Distortion measurement Nuclear magnetic resonance Signal to noise ratio Time domain analysis Frequency estimation Noise measurement

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Towards optimal quantization in multistage audio coding

Towards optimal quantization in multistage audio coding

引用

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

作者： R. Vafin W.B. Kleijn Speech Processing Group Department of Signals Sensors and Systems KTH Royal Institute of Technology Stockholm Sweden

We develop a new method for quantization in multistage audio coding. We consider the case of a two-stage sinusoidal/waveform coder. Given a distortion measure and a bit-rate constraint, we analytically derive the optimal rate distribution between subcoders (stages) and the corresponding optimal quantizers, which allows the coder to adapt easily to changes in bit-rate requirements. We verify that the performance, both in terms of signal-to-noise ratio (SNR) and perceptual quality, is higher if the input to the second stage is obtained by subtracting the quantized first-stage reconstruction from the original signal, as opposed to subtracting the unquantized reconstruction.

关键词： Quantization audio coding Distortion measurement Bit rate Speech enhancement Speech processing Signal processing Sensor systems Signal to noise ratio Speech coding

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Extending Fine-Grain Scalable audio coding to Very Low Bitrates using Overcomplete Dictionaries

Extending Fine-Grain Scalable Audio Coding to Very Low Bitra...

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Emmanuel Ravelli Gael Richard Laurent Daudet Institut Jean Le Rond d'Alembert LAM Team University Pierre et Marie Curie Paris France TSI Department GET/ENST (Télécom Paris) Paris France Institut Jean Le Rond d'Alembert LAM Team University Pierre et Marie Curie-Paris 6 Paris France

Signal representations in overcomplete dictionaries are considered here as an alternative to the traditional transform representations for fine-grain scalable audio coding. Such representations produce sparser decompositions and thus allow better coding efficiency than transform coding at very low bitrates. Moreover, the decomposition algorithms are intrinsically progressive, and flexible enough to allow an efficient transient modeling. We propose in this paper a fine-grain scalable audio coder which works on a large range of bitrates (2kbs to 128kbs). Objective measures as well as informal subjective evaluation show that this coder outperforms a comparable transform-based coder at very low bitrates.

关键词： audio coding Bit rate Dictionaries MPEG 4 Standard Transform coding Image coding Bandwidth Signal representations Time frequency analysis Conferences

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Improving joint stereo audio coding by adaptive inter-channel prediction

Improving joint stereo audio coding by adaptive inter-channe...

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： H. Fuchs Institut für Theoretische Nachrichtentechnik und Infomtionsverarbeitung Universität Hannover Hanover Germany

A method for exploiting inter-channel redundancies of stereophonic or multichannel audio signals is presented. In contrast to known stereo redundancy reduction techniques used in joint stereo audio coding. Where only the statistical dependencies between two concurrent samples of the left and right channel signals are considered, the adaptive inter-channel prediction also takes into account possible phase or time delay between the channels and exploits more than only one value of the cross-correlation function. The analysis of subjective listening test results has shown that this technique is especially effective for a class of test sequences which has proven to be most critical for the ISO MPEG Layer II and Layer III codecs at bit rates of 2/spl times/64 kbit/s. For these signals the gain due to the stereo redundancy reduction technique used in Layer III joint stereo coding is less than 5-10 dB, while in Layer II joint stereo coding no specific stereo redundancy reduction technique is used. In a first step, the adaptive inter-channel prediction has been applied to an ISO MPEG Layer II codec. The simulation results show that a prediction gain up to 30-40 dB can be achieved for large parts of the above mentioned signals.< >

关键词： audio coding Testing Bit rate Codecs Delay effects Distortion Instruments

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ITU-T G.722.1 Annex C: A New Low-Complexity 14 KHZ audio coding Standard

ITU-T G.722.1 Annex C: A New Low-Complexity 14 KHZ Audio Cod...

引用

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

作者： Minjie Xie D. Lindbergh P. Chu Polycom Inc. Burlington MA USA Polycom Inc. Andover MA USA

This paper describes the low-complexity 14 kHz audio coding algorithm which has been recently standardized by ITU-T as Recommendation G.722.1 Annex C ("G.722.1C"). The algorithm is an extension to ITU-T Recommendation G.722.1 and doubles the G.722.1 algorithm to permit 14 kHz audio bandwidth using a 32 kHz audio sample rate, at 24, 32, and 48 kbit/s. The G.722.1C codec features very high audio quality and extremely low computational complexity compared to other state-of-the-art audio coding algorithms. This codec is suitable for use in video conferencing and teleconferencing, and Internet streaming applications. Subjective test results from the characterization phase of G.722.1 C are also presented in the paper

关键词： audio coding Codecs Videoconference Teleconferencing Bandwidth Testing Wideband Video sharing Speech Computational complexity

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Error-Protected TwinVQ audio coding at Less Than 64 kbit/s/ch

Error-Protected TwinVQ Audio Coding at Less Than 64 kbit/s/c...

引用

IEEE Workshop on Speech coding For Telecommunications

作者： K. Ikeda T. Moriya N. Iwakami S. Miki NTT Human Interface Laboratories Musashino Tokyo Japan

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