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

作者： Bernd Geiser Hauke Kruger Heinrich W. Lollmann Peter Vary Deming Zhang Hualin Wan Hai Ting Li Li Bin Zhang Institute of Communication Systems and Data Processing (ind) RWTH Aachen University Germany Core Network Research Dept. Huawei Technologies Co. Ltd. Beijing P.R. of China

This paper describes the speech and audio codec that has been submitted to ITU-T by Huawei and ETRI as a candidate for the upcoming super-wideband and stereo extensions of Rec. G.729.1 and G.718. The core codec in the current implementation is G.729.1 and the encoded frequency range is increased from 7 kHz to 14 kHz. Therefore, the maximum bit rate is raised from 32 kbit/s to 64 kbit/s by adding five bitstream layers. A comprehensive overview of the codec is presented with a focus on the mono coding components. The results of the listening tests that have been conducted during the ITU-T qualification phase are summarized. The proposed codec passes all quality requirements for mono input signals.

关键词： Proposals Speech coding audio coding Codecs MONOS devices Decoding Sampling methods Bandwidth Bit rate Filter bank

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A phylogenetic analysis of near-duplicate audio tracks

A phylogenetic analysis of near-duplicate audio tracks

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IEEE Workshop on Multimedia Signal Processing

作者： Matteo Nucci Marco Tagliasacchi Stefano Tubaro Dipartimento di Elettronica P.zza Leonardo da Vinci Milano Italy

We present a content-based system for the analysis of near-duplicate audio tracks. The objective is to infer the structure of modifications, represented as trees, underneath a pool of near-duplicates, possibly specifying the operations the tracks have gone through. A pilot study was carried out for a set of plausible processing operators, including trim, fade and perceptual audio coding, generating near-duplicates from an original audio track. The proposed method measures the similarity between pairs of near-duplicates and reconstructs a tree representing the causal dependencies in the analyzed pool. Experimental results demonstrate that the structure of the tree can be successfully recovered, also in the challenging case in which some pieces of information are missing, i.e., when only a subset of near-duplicate tracks is available.

关键词： Vegetation Signal to noise ratio Measurement Phylogeny audio coding Reconstruction algorithms Object recognition

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Index assignment-based channel coding

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IET COMMUNICATIONS 2012年第2期6卷 172-178页

作者： Oztoprak, H. Villette, S. Kondoz, A. Univ Surrey Ctr Vis Speech & Signal Proc Guildford GU2 7XH Surrey England

A scheme named index assignment-based channel coding (IACC) has been developed for resilience of speech/audio codecs against the bit errors commonly experienced in wireless channels. Although IACC is a type of joint source channel coding, it does not intervene with the source codec design. The proposed scheme takes into account source characteristics and adjusts the amount of coding according to the sensitivity of different values of the source parameters. It is shown that source characteristics play an important role in the performance of IACC. The performance of IACC has been evaluated by applying it to parameters generated by adaptive multi-rate wideband (AMR-WB+) audio codec. A method for perceptual training of IACC codes is also proposed. The results demonstrate that the performance of IACC and IACC concatenated with convolutional coding can be superior to that of conventional convolutional coding at high and moderate bit error rates, respectively.

关键词： combined source-channel coding convolutional codes error statistics source codec design wireless channels concatenated coding adaptive multirate wideband audio codec index assignment-based channel coding speech coding bit error rates Speech and audio coding audio coding adaptive codes concatenated codes speech codecs joint source channel coding Speech processing techniques BER Radio links and equipment convolutional coding Other topics in statistics

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Parametric AM/FM decomposition for speech and audio coding

Parametric AM/FM decomposition for speech and audio coding

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Tom Bäckström Sascha Disch Fraunhofer Institute of Integrated Circuits and Systems Erlangen Germany Leibniz Universität Hannover Hanover Germany

audio transform coding methods generally employ an assumption of locally stationary signals. However, as demands on coding efficiency increase, the assumption of stationarity becomes increasingly challenging. In this article we study a parametric method for modeling variability in an audio signal, especially with voiced speech in mind. Namely, we model modulation in fundamental frequency and amplitude, with the objective of allowing normalization of the signal to satisfy the stationarity requirements of transform based coders.

关键词： Speech coding audio coding Frequency modulation Transform coding Noise shaping Amplitude modulation Conferences Acoustic signal processing Acoustic applications Speech processing

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A MDCT BASED HARMONIC SPECTRAL BANDWIDTH EXTENSION METHOD

A MDCT BASED HARMONIC SPECTRAL BANDWIDTH EXTENSION METHOD

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Christian Neukam Frederik Nagel Gerald Schuller Michael Schnabel Fraunhofer IIS Am Wolfsmantel 33 91058 Erlangen Germany Ilmenau University of Technology Ehrenbergstr. 29 98693 Ilmenau Germany

ISBN: (纸本)9781479903573

Modern audio coding technologies apply methods of bandwidth extension (BWE) to efficiently represent audio data at low bitrates. An established method is the well-known spectral band replication (SBR) that is part of MPEG High Efficiency Advanced audio coding (HE-AAC). However, if the signal features a distinct harmonic spectral structure, the use of these methods tends to result in audible artifacts, because the harmonic structure is not reconstructed correctly. In this paper a bandwidth extension method is proposed which eliminates the undesirable effects and allows for an efficient implementation in the Modified Discrete Cosine Transform (MDCT) domain. The proposed Harmonic Spectral Bandwidth Extension (HSBE) method uses arbitrary frequency shifts for modulating the replicated spectrum in a way that the harmonic structure of the signal is preserved. A listening test demonstrates the advantage of the proposed method compared to the state of the art.

关键词： Bandwidth extension audio coding MDCT MDST audio coding mdct(modified discrete cosine transform) harmonics Bandwidth state of the art audio data undesirable effects

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Temporal quantization of spatial information using directional clustering for multichannel audio coding

Temporal quantization of spatial information using direction...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Shigeki Miyabe Keisuke Masatoki Hiroshi Saruwatari Kiyohiro Shikano Toshiyuki Nomura University of Tokyo Tokyo Japan Nara Institute of Science and Technology Nara Japan NEC Corporation Limited Kanagawa Japan

Binaural cue coding, which is a representing low bit-rate coding of multichannel audio, generates large distortion when the audio data have complex spatial image, such as symphony. Such distortion caused by the low frequency resolution of spatial information because BCC quantizes the parameters of localization. In this paper we propose a new coding framework by quantizing the spatial information temporally. The single-channel sum signal is panned to the multiple channels by selecting the prototypes of the spatial filter. Optimization of the prototypes with minimum coding error is given by a k-means-like clustering of the angles whose centroids are given by the first principal components of the covariances in the classes. The efficiency of the proposed coding with high quality is verified both in the objective and subjective evaluations.

关键词： Quantization audio coding Signal generators Frequency Signal resolution Prototypes Filter bank Image coding Acoustic distortion Spatial resolution

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SINUSOIDAL COMPONENT SELECTION BASED ON PARTIAL LOUDNESS CRITERIA

SINUSOIDAL COMPONENT SELECTION BASED ON PARTIAL LOUDNESS CRI...

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IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Harish Krishnamoorthi Andreas Spanias School of Electrical Computer and Energy Engineering SenSIP Center Arizona State University Tempe AZ USA 85287-5706

ISBN: (纸本)9781479903573

Sinusoidal models are widely used in parametric speech and audio coding schemes. A common requirement in these applications is to select only a subset of components that provide the greatest perceptual benefit particularly at low bit-rates. Usually, perceptual sinusoidal component selection algorithms make use of greedy algorithms that are computationally expensive. In this paper, we present a new algorithm that selects sinusoidal components based on the partial loudness model proposed by Moore & Glasberg. We compare the performance of the proposed algorithm in terms of perceptual benefit and computational complexity to other existing sinusoidal selection algorithms.

关键词： loudness sinusoidal models parametric audio coding audio coding auditory patterns

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Temporal Transient Localization for Enhanced Pre-Echo Suppression

Temporal Transient Localization for Enhanced Pre-Echo Suppre...

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2013 Asian Pacific Conference on Mechatronics and Control Engineering(APCMCE 2013)

作者： Wenhua Ma Jin Xu Yuli You School of Informatics Guangdong University of Foreign Studies

A specially designed short window of reduced effective size and other supporting windows,as used in a switched MDCT audio codec,are sequenced in such a way that this special window is placed around transient samples,thereby providing transient localization while leaving frequency resolution unchanged. Pre-echo is better suppressed because transient localization reduces thespread of both quantization noise and high bitrates associated with *** overheads include one or two additiona lbits and a short procedure,both for windows equencing,and more memory for storing additional window functions.

关键词： audio coding transient pre-echo window switching window size

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Rate-distortion optimization for multichannel audio compression

Rate-distortion optimization for multichannel audio compress...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Minyue Li Jan Skoglund W. Bastiaan Kleijn Google Inc. Mountain View CA

Multichannel audio coding is studied from a rate-distortion theoretical viewpoint. Two practical coding techniques, both of which are based on rate-distortion optimization, are also proposed. The first technique decorrelates a multichannel signal hierarchically using elementary unitary transforms. The second method rearranges a multichannel signal into sub-signals and compresses them at optimized bit rates using a conventional codec. Both objective and subjective tests were conducted to illustrate the efficiency of the methods.

关键词： Transforms Codecs Decorrelation Resource description framework Resource management audio coding

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audio codingwith power spectral density preserving quantization

Audio codingwith power spectral density preserving quantizat...

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

作者： Minyue Li Janusz Klejsa Alexey Ozerov W. Bastiaan Kleijn School of Electrical Engineering KTH Royal Institute of Technology Stockholm Sweden Technicolor Inc. Cesson-Sevigne France School of Engineering and Computer Science Victoria University of Wellington New Zealand

The coding of audio-visual signals is generally based on different paradigms for high and low rates. At high rates the signal is approximated directly and at low rates only signal features are transmitted. The recently introduced distribution preserving quantization (DPQ) paradigm provides a seamless transition between these two regimes. In this paper we present a simplified scheme that preserves the power spectral density (PSD) rather than the probability distribution. In a practical system the PSD must be estimated. We show that both forward adaptive and backward adaptive PSD estimation are possible. Our experimental results confirm that preservation of PSD at finite precision leads to a unified coding paradigm that provides effective coding at both high and low rates. An audio coding application shows the perceptual benefits of PSD preserving quantization.

关键词： Quantization audio coding Estimation Adaptation models Lattices Speech

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