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Low delay LPC and MDCT-based audio coding in the EVS codec

Low delay LPC and MDCT-based audio coding in the EVS codec

引用

IEEE International Conference on Acoustics, Speech and Signal Processing

作者： G. Fuchs C. R. Helmrich G. Markovic M. Neusinger E. Ravelli T. Moriya Fraunhofer Inst. fur Integrierte Schaltungen (IIS) Erlangen Germany

ISBN: (纸本)9781467369985

Speech coders operating in time domain can be extended with a frequency domain mode to improve encoding of music, even though this is challenging at low delay. In such a scenario, the short analysis window limits the benefit of the transform coder, while a delayless switch between the two coders constrains the system further. The paper presents an LPC and MDCT-based audio coder part of the new 3GPP codec for Enhanced Voice Services, which aims to solve the issues. Several advanced coding tools are introduced to alleviate the constraints: transient handling is improved, harmonic structures are better preserved, and the modeling of the zero-quantized frequencies is enhanced. Test results show that the obtained low-delay switched coder brings a clear improvement over a speech coder and is competitive even in comparison to audio coders with higher delay.

关键词： audio coding EVS LPC Low delay MDCT

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Wideband speech and audio coding using gammatone filter banks

Wideband speech and audio coding using gammatone filter bank...

引用

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

作者： E. Ambikairajah J. Epps L. Lin School of Electrical Engineering and Telecommunications The University of New South Wales NSW Australia School of Electrical Engineering and Telecommunications The University of New South Wales Sydney NSW Australia

Considerable research attention has been directed towards speech and audio coding algorithms capable of producing high quality coded speech and audio, however few of these use signal representations which account for temporal as well as spectral detail. This paper presents a new technique for 16 kHz wideband speech and audio coding, whereby analysis and synthesis are performed using a linear phase gammatone filter bank. The outputs of these critical band filters are processed to obtain a series of pulse trains that represent neural firing. Auditory masking is then applied to reduce the number of pulses, producing a more compact time-frequency parameterization. The critical band gains and pulse amplitudes and positions are then coded using a combination of non-uniform quantization, arithmetic coding and vector quantization. This coding paradigm produces high quality coded speech and audio, is based upon well-known models of the auditory system, is highly scalable, and has moderate complexity.

关键词： Wideband Speech coding audio coding Filter bank Speech synthesis Speech analysis Signal representations Performance analysis Signal synthesis Time frequency analysis

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FPGA prototyping of an in-situ reconfigurable filter bank for audio coding

FPGA prototyping of an in-situ reconfigurable filter bank fo...

引用

Midwest Symposium on Circuits and Systems (MWSCAS)

作者： L.A. de B Naviner J.-F. Naviner M.A. de Barros Centre National de la Recherche Scientifique Paris ÃŽle-de-France FR GET/Télécom Paris CNRS/LTCI Paris France Universidade Federal de Campina Grande Departamento de Sistemas deComputação Campina Grande Brazil

This paper presents the FPGA implementation of a processor suitable for adaptive multi-resolution analysis in a audio coding schema. Implementation takes into account all symmetries in the filter response and uses distributed arithmetic approach in order to reduce the device cells account and routing requirements. The prototyped circuit allows dynamic reconfiguration of subbands number and width

关键词： Field programmable gate arrays Prototypes Filter bank audio coding Psychoacoustic models Humans Psychology Masking threshold Arithmetic Routing

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Globally optimized dynamic bit-allocation strategy for subband ADPCM-based low delay audio coding

Globally optimized dynamic bit-allocation strategy for subba...

引用

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

作者： Stephan Preihs Jörn Ostermann Leibniz Universität Hannover Institut für Informationsverarbeitung Appelstr. 9A Hannover Germany

This paper presents an extension and global optimization of a bit-allocation strategy for use in a low delay subband ADPCM-based audio coding scheme. The framework employed for the assignment of bits to subband quantizers is based on a subband level estimation of the signals used for the prediction error normalization. It is modified to reduce switching artifacts and extended to contain a mapping to predefined allocation presets. Since our bit-allocation scheme as well as the subband coding involve several partially interacting parameters that are hard to adjust manually, a framework for their global optimization is presented. Experiments and their results show that in our coding scheme a significant improvement of audio quality without large signaling overhead can be achieved by the use of the modified bit-allocation. Furthermore, the global optimization allows for an additional gain in audio quality compared to manually adjusted parameters.

关键词： Delays Optimization Codecs Resource management audio coding Dynamic scheduling

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A new approach to low bit rate audio coding using a combined harmonic-multiband-wavelet representation

A new approach to low bit rate audio coding using a combined...

引用

International Symposium on Signal Processing and Its Applications (ISSPA)

作者： M. Deriche D. Ning S. Boland Signal Processing Research Centre School of Electrical and Electronic Systems Engineering Queensland University of Technology Brisbane QLD Australia

In this paper, we present updated results for the combined harmonic-wavelet based audio coder. This coder first approximates the audio frame as a sum of several sinusoids using a harmonic analysis-synthesis scheme. The reconstructed harmonic signal is then subtracted from the original to give a residual, which is analysed using a wavelet filtering scheme. After each step (harmonic analysis and wavelet filtering), the perceptually relevant parameters are quantised and encoded. The total least squares (TLS)-Prony algorithm is used for the harmonic analysis scheme, and cascaded M-band wavelet transforms are used for analysing the residual. At encoding bit rates of 60-70 kbit/s, this audio coder is capable of delivering better results than the MPEG layer II audio coder operating at the much higher bit rate of 128 kbit/s.

关键词： Bit rate audio coding Harmonic analysis Wavelet analysis Power harmonic filters Filtering Least squares approximation Signal analysis Least squares methods Wavelet transforms

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A modified distortion metric for audio coding

A modified distortion metric for audio coding

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

作者： Vinay Melkote Kenneth Rose Department of Electrical and Computer Engineering University of California Santa Barbara CA USA

ISBN: (纸本)9781424423538

Current audio coding standards employ the modified discrete cosine transform (MDCT) where overlapped frames of audio are windowed and transformed to the frequency domain. Encoding parameters are chosen so as to minimize a distortion measure subject to a rate constraint. At the decoder, inverse transformation involves additional windowing and overlap-add of frames. An analysis of the time domain error in the reconstructed frame reveals that distortion metrics based solely on the MDCT domain error are in fact unable to capture the effects of windowing and overlap-add at the decoder. The main contribution of this paper is a modified distortion metric that does capture these effects via modified discrete sine transform analysis. When incorporated into an Advanced audio Coder the proposed distortion metric significantly improves subjective quality of reconstructed audio.

关键词： audio coding Decoding Discrete transforms Quantization Distortion measurement Frequency domain analysis Discrete Fourier transforms Discrete cosine transforms Time domain analysis Entropy

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Cascaded RLS-LMS Prediction in MPEG-4 Lossless audio coding

Cascaded RLS-LMS Prediction in MPEG-4 Lossless Audio Coding

引用

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

作者： H. Huang S. Rahardja X. Lin R. Yu P. Franti Institute for Infocomm Research Singapore Department of Computer Science University of Joensuu Finland

A new MPEG-4 standard for lossless audio coding is going to be published in 2006. This coming international standard consists of two parts: the transform-domain scalable to lossless coding (SLS), and the time-domain audio lossless coding (ALS). In ALS, linear prediction is used to compress the dynamic ranges of the input audio signal. The prediction residual is coded by an entropy coder with either Rice code or arithmetic code. There are two prediction modes in ALS: linear predictive coding (LPC) and cascaded RLS-LMS. As the developer of the RLS-LMS prediction, we present this technology in this paper. In RLS-LMS prediction, the input audio samples go through the cascaded DPCM, RLS, and LMS predictors, whose output predictions are linearly combined to generate a prediction for the current input sample. Through MPLG testings, it has been found that ALS with RLS-LMS prediction provides the best lossless compression ratio compared with SLS, ALS with LPC, and several non-MPEG codecs

关键词： MPEG 4 Standard audio coding Linear predictive coding Laser sintering Standards publication Time domain analysis Dynamic range Entropy Arithmetic Resonance light scattering

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Low bit rate high quality audio coding with combined harmonic and wavelet representations

Low bit rate high quality audio coding with combined harmoni...

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

作者： K.N. Hamdy M. Ali A.H. Tewfi Department of Electrical Engineering University of Minnesota Minneapolis MN USA University of Minnesota Minneapolis MN US

We describe a novel high quality audio coding method using adaptive signal representation, based on sinusoidal and wavelet analysis of signals. First, we perform a harmonic analysis of the signal to remove strong periodic structures or tones from the signal. Then we carry out wavelet analysis that are useful in tracking the transients of the signal. These transients are then removed from the wavelet coefficients. The remaining coefficients have broadband noise-like structure. Since this method separates out tones (sinusoids), transients, and broadband noise, we may use tonal, noise, and temporal masking information to individually encode the tones and the wavelet coefficients. Our experiments suggest that this method yields a nominal bit rate of 1 bit/sample for high quality audio compression.

关键词： Bit rate audio coding Wavelet analysis Signal analysis Wavelet coefficients Signal representations Harmonic analysis Periodic structures Transient analysis audio compression

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Lossless audio coding with Bandwidth Extension Layers

Lossless Audio Coding with Bandwidth Extension Layers

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Stephen Voran Institute of Telecommunication Sciences Boulder CO USA

Layered audio coding typically offers reduced distortion as bit rate is increased, but that distortion is spread across the entire band until the lossless coding bit rate is reached and distortion is eliminated. We propose a layered audio coding paradigm of bandwidth extension, rather than distortion reduction. For example, a core layer can provide lossless coding of a 24 kHz bandwidth signal (f s =48 kHz), then first and second bandwidth extension lossless layers can extend that signal to losslessly coded 48 and then 96 kHz bandwidths (f s =96 and 192 kHz).

关键词： audio coding Bandwidth Finite impulse response filter Acoustic distortion Band pass filters Bit rate Signal resolution Passband Sampling methods Frequency

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Jointly optimal quantization of parameters in sinusoidal audio coding

Jointly optimal quantization of parameters in sinusoidal aud...

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： R. Vafin W.B. Kleijn Department of Signals Royal Institute of Technology Stockholm Sweden

In this paper, we present new methods for jointly optimal scalar quantization of parameters in sinusoidal audio coding. The sinusoidal amplitudes, frequencies, and phases used to represent (the tonal component of) an audio signal are quantized such as to minimize a single-letter perceptual distortion measure under a given bit-rate constraint. The asymptotically optimal scalar quantizers are derived analytically using high-rate theory. We present different practical methods, where quantization accuracy of some parameters is made dependent on quantized values of other parameters. The choice of method depends on the availability of the frequency-masking threshold, defining the perceptual importance of the individual sinusoids, at the decoder. The high performance of our quantizers is demonstrated by means of a listening test

关键词： Quantization audio coding Distortion measurement Decoding Bit rate Frequency estimation Entropy Sensor systems Phase distortion Frequency measurement

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