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Algorithm for achieving adaptive tiling of time axis for audio coding purposes

ELECTRONICS LETTERS 2002年第9期38卷 434-435页

作者： Ruiz, N Rosa, M López, F Vera, P Univ Jaen Dept Elect Escuela Politecn Linares Jaen Spain Univ Alcala de Henares Escuela Politecn Dept Teoria Senal & Commun Madrid Spain

A new algorithm for achieving flexible tiling, of the time axis for audio coding purposes is presented, It is based on the calculus of the distances among a predetermined number of time-frequency pairs, From the computed distances. a clustering process determines the final subdivision of each audio frame. Experimental results demonstrates the good performance of the proposed algorithm. which provides high coding, efficiency with a reduced complexity.

关键词： audio coding clustering process audio frame complexity reduction adaptive tiling time-frequency pairs Speech and audio coding time axis high coding efficiency adaptive signal processing

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Quantization of wavelet packet audio coding

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High Technology Letters 2006年第3期12卷 295-299页

作者：谭建国 Zhang Wenjun LiuPeilin Institute of Image Communication and Information Processing Shanghai Jiaotong University Shanghai 200240 P.R. China

Abstract The method of quantization noise control of audio coding in the wavelet domain is proposed. Using the inverse Discrete Fourier Transform （DFT）, it converts the masking threshold coming from MPEG psycho-acoustic model in the frequency domain to the signal in the time domain; the Discrete Wavelet Packet Transform （DWPF） is performed; the energy in each subband is regarded as the maximum allowed quantization noise energy. The experimental result shows that the proposed method can attain the nearly transparent audio quality below 64kbps for the most testing audio signals.

关键词： quantization wavelet packet audio coding DFT

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MPEG-H 3D audio: Immersive audio coding

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ACOUSTICAL SCIENCE AND TECHNOLOGY 2022年第2期43卷 143-148页

作者： Herre, Jorgen Quackenbush, Schuyler R. Int Audio Labs Erlangen Erlangen Germany Fraunhofer IIS Erlangen Germany Audio Res Labs Scotch Plains NJ 07076 USA

The term "Immersive audio"is frequently used to describe an audio experience that provides to the listener the sensation of being fully immersed or "present"in a sound scene. This can be achieved via different presentation modes, such as surround sound (several loudspeakers horizontally arranged around the listener), 3D audio (with loudspeakers at, above and below listener ear level) and binaural audio to headphones. This article provides an overview of the recent MPEG standard, MPEG-H 3D audio, which is a versatile standard that supports multiple immersive sound signal formats (channels, objects, higher order ambisonics), and is now being adopted in broadcast and streaming applications.

关键词： 3D audio audio coding audio compression audio data reduction Immersive audio MPEG MPEG-H

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Rate-distortion optimized quantization in multistage audio coding

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IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2006年第1期14卷 311-320页

作者： Vafin, R Kleijn, WB Royal Inst Technol KTH Dept Signals Sensors & Syst Tallinn Estonia Royal Inst Technol Dept Signals Sensors & Syst S-10044 Stockholm Sweden

In this work, we develop a new method for quantization in multistage audio coding. Given a (perceptual) distortion measure and a bit-rate constraint, we analytically derive the optimal rate distribution between subcoders (stages) and the corresponding optimal quantizers using high-rate theory. The analytical solutions for optimal quantizers allow a coder to easily adapt to changes in bit-rate requirements. As an illustration of the new method, we consider quantization in a two-stage sinusoidal/wave form coder that is a widely used combination in audio coding. We show that at low total rates most of the rate should be assigned to the sinusoidal (model-based, subspace) subcoder, while at high total rates most of the rate should be assigned to the waveform (full-space) subcoder. We compare the new method to a reference quantization method that does not use rate-distortion optimization. A significantly higher performance of the new method is shown by means of a listening test.

关键词： audio coding high-rate theory modified discrete cosine transform (MDCT) multistage coding quantization rate-distortion optimization sinusoidal coding waveform coding

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Transform audio coding with arithmetic-coded scalar quantization and model-based bit allocation

Transform audio coding with arithmetic-coded scalar quantiza...

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32nd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： 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)1424407281

In this paper we present a new model-based method to code the transform coefficients of audio signals. The histogram of transform coefficients is approximated by a generalized Gaussian model for efficient model-based bit allocation and the spectrum is coded by scalar quantization followed by arithmetic coding. An example coder operating at 16 kHz and using predictive modified discrete cosine transform (MDCT) coding is described. We compare the performance of the proposed coder with ITU-T G.722.1. Objective and subjective quality results are presented. The proposed coder is better than ITU-T G.722.1 at 24 kbit/s and equivalent at 32 kbit/s.

关键词： transform coding audio coding modeling

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Joint filterbank structures for integrating audio coding into hearing aid applications

Joint filterbank structures for integrating audio coding int...

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

作者： Dong, Rong Hermann, David Brennan, Robert Chau, Edward AM Semicond Waterloo ON N2V 1K8 Canada

ISBN: (纸本)9781424414833

To help develop ultra-low power wireless hearing aid products, we investigate the integration of subband audio coding with hearing aid applications. Both the audio coding and the hearing aid application use subband processing, but their requirements for the filterbanks are totally different. The hearing aid application typically uses an oversampled filterbank to reduce the aliasing in each subband, whereas the audio codec needs a critically sampled filterbank for maximal coding efficiency. A joint filterbank structure is proposed in this paper to satisfy these contradictive filterbank requirements. With this structure, the two filterbanks are combined into a single stereo filterbank operation, which can be efficiently implemented. on a filterbank coprocessor. This structure substantially reduces the computation complexity, power consumption and memory usage.

关键词： channel bank filters transform coding audio coding hearing aids coprocessors

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SIGNAL-ADAPTIVE TRANSFORM KERNEL SWITCHING FOR STEREO audio coding

SIGNAL-ADAPTIVE TRANSFORM KERNEL SWITCHING FOR STEREO AUDIO ...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics (WASPAA)

作者： Helmrich, Christian R. Edler, Bernd Univ Erlangen Nurnberg Int Audio Labs Erlangen Wolfsmantel 33 D-91058 Erlangen Germany

ISBN: (纸本)9781479974504

Modern stereo and multi-channel perceptual audio codecs utilizing the modified discrete cosine transform (MDCT) can achieve very good overall coding quality even at low bit-rates but lack efficiency on some material with inter-channel phase difference (IPD) of about +/-90 degrees. To address this issue a generalization of the lapped transform coding scheme is proposed which retains the perfect reconstruction property while allowing the usage of three further transform kernels, one of which is the modified discrete sine transform (MDST). Blind listening tests indicate that by frame-wise adaptation of each channel's transform kernel to the instantaneous IPD characteristics, notable gains in coding quality are possible with only negligible increase in decoder complexity and parameter rate.

关键词： audio coding joint stereo MDCT MDST

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Dual-band PWM audio coding for Ultrasound Artefact Reduction 25

Dual-band PWM Audio Coding for Ultrasound Artefact Reduction

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25th European Signal Processing Conference (EUSIPCO)

作者： Kaleris, Konstantinos Mourjopoulos, John Univ Patras Elect & Comp Engn Dept Audio & Acoust Technol Grp Patras 26500 Greece

ISBN: (纸本)9780992862671

A novel approach to PWM coding is introduced based on generating two complementary PWM streams out of the in-band signal's spectrum. The 2 streams are then recombined in a suitable way, so that out-of-phase cancellation of the carrier frequency harmonics is achieved. The approach suppresses the strong out-of-band frequencies of the carrier signal without introducing distortion of the in-band coded signal. Such method can achieve superior reduction compared to the out-of-band artefact suppression induced by traditional analog low-pass filters employed in typical Class-D audio amplifiers or other switching power delivery systems, hence allowing designs with reduced filter requirements or even filterless implementations.

关键词： Pulse Width Modulation Class-D audio amplifier audio coding PWM ultrasound

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IMMERSIVE audio coding FOR VIRTUAL REALITY USING A METADATA-ASSISTED EXTENSION OF THE 3GPP EVS CODEC 44

IMMERSIVE AUDIO CODING FOR VIRTUAL REALITY USING A METADATA-...

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44th IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)

作者： McGrath, D. Bruhn, S. Purnhagen, H. Eckert, M. Torres, J. Brown, S. Darcy, D. Dolby Australia Pty Ltd Sydney NSW Australia Dolby Sweden AB Stockholm Sweden Dolby Labs Inc San Francisco CA USA

ISBN: (纸本)9781479981311

Virtual Reality (VR) audio scenes may be composed of a very large number of audio elements, including dynamic audio objects, fixed audio channels and scene-based audio elements such as Higher Order Ambisonics (HOA). Potentially, the subjective listening experience may be replicated using a compact spatial format with a set number of dynamic objects and scene-based elements, retaining only the perceptual essence of the audio scene. The compact format would further enable a reduction in the complexity of subsequent compression and rendering. This paper investigates these hypotheses by exploring the use of a compact format that consists of up to four dynamic objects and nine HOA channels, with the Enhanced Voice Services (EVS) codec being applied to a 4-channel down-mix of the compact format.

关键词： audio coding Virtual Reality Spatial audio Immersive audio Ambisonics

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A ROBUST SPEECH/MUSIC DISCRIMINATOR FOR SWITCHED audio coding 23

A ROBUST SPEECH/MUSIC DISCRIMINATOR FOR SWITCHED AUDIO CODIN...

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23rd European Signal Processing Conference (EUSIPCO)

作者： Fuchs, Guillaume Fraunhofer Inst Integrierte Schaltungen IIS Erlangen Germany

ISBN: (纸本)9780992862633

Switching between speech coding and generic audio coding schemes was recently proven to be very efficient for coding a large range of audio materials at low bit-rates. However, it strongly relies on a robust classification of the input signal. The aim of the paper is to design a reliable speech and music discriminator (SMD) for such an application. Main attention was laid on getting a good tradeoff between accuracy, reactivity and stability of the decision while keeping the delay and complexity reasonably low. To this end, short-term and long-term features are dissociated before being conveyed to two different classifiers. The two classifier outputs are combined in a final decision using a hysteresis. Objective measures show that a more reliable switching decision is achievable. The SMD was successfully implemented in MPEG Unified Speech and audio coding (USAC). It allows the codec to show unprecedented audio quality.

关键词： Speech and Music-Discrimination Speech coding audio coding

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