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Computational improvement for G.729 standard

ELECTRONICS LETTERS 2000年第13期36卷 1163-1164页

作者： Hwang, SH Ming Hsin Inst Technol Dept Management Informat Hsinchu Taiwan

G.729 is a CS-ACELP-based [1] standard for 8kbit/s speech compression and is defined by the ITU. In the G.729 encoding phase. the open-loop pitch T-op is estimated for each frame to reduce the computation required to search for the best adaptive codebook delay. However, > 25% CPU time is required for the open-loop pitch analysis. The weighted delta-LSP(WD-LSP) function is employed here to reduce the computation required for open-loop pitch analysis. An 81% saving is made in the computation of T-op. A total computational saving of 21% for the encoding phase Is achieved and the speech quality still sounds good (MOS = 3.5).

关键词： telecommunication standards open-loop pitch data compression linear predictive coding encoding phase speech quality CS-ACELP-based standard G.729 standard Codes speech coding open-loop pitch analysis 8 kbit/s speech and audio coding ITU computational saving speech compression weighted delta-LSP adaptive codebook delay

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Voice activity detector based on CAPDM architecture

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ELECTRONICS LETTERS 2001年第1期37卷 68-69页

作者： Liu, CH Huang, CC Natl Chiao Tung Univ Dept Commun Engn Wireless Commun Lab Hsinchu Taiwan

A voice activity detector based on the controlled adaptive prediction delta modulation architecture is presented. The proposed detector utilises two parameters, a prediction gain and a prediction indicator. Computer s... 详细信息

关键词： adaptive signal detection speech processing techniques Signal detection voice activity detector voice activity detection algorithm controlled adaptive prediction delta modulation prediction gain Modulation and coding methods CAPDM architecture prediction theory speech coding delta modulation noisy environment speech and audio coding Signal processing theory robust algorithm prediction indicator

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Greedy sparse decompositions: a comparative study

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EURASIP JOURNAL ON ADVANCES IN SIGNAL PROCESSING 2011年第1期2011卷 1页

作者： Dymarski, Przemyslaw Moreau, Nicolas Richard, Gael Warsaw Univ Technol Inst Telecommun Warsaw Poland CNRS LTCI Inst Telecom Telecom ParisTech Paris France

The purpose of this article is to present a comparative study of sparse greedy algorithms that were separately introduced in speech and audio research communities. It is particularly shown that the Matching Pursuit (MP) family of algorithms (MP, OMP, and OOMP) are equivalent to multi-stage gain-shape vector quantization algorithms previously designed for speech signals coding. These algorithms are comparatively evaluated and their merits in terms of trade-off between complexity and performances are discussed. This article is completed by the introduction of the novel methods that take their inspiration from this unified view and recent study in audio sparse decomposition.

关键词： greedy sparse decomposition matching pursuit orthogonal matching pursuit speech and audio coding

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Spectral Noise Shaping: Improvements in speech/audio Codec Based on Linear Prediction in Spectral Domain

Spectral Noise Shaping: Improvements in Speech/Audio Codec B...

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9th Annual Conference of the International-speech-Communication-Association (INTERspeech 2008)

作者： Ganapathy, Sriram Motlicek, Petr Hermansky, Hynek Garudadri, Harinath IDIAP Res Inst Martigny Switzerland Ecole Polytech Fed Lausanne Lausanne Switzerland Qualcomm Inc San Diego CA USA

ISBN: (纸本)9781615673780

audio coding based on Frequency Domain Linear Prediction (FDLP) uses auto-regressive models to approximate Hilbert envelopes in frequency sub-bands. Although the basic technique achieves good coding efficiency, there is a need to improve the reconstructed signal quality for tonal signals with impulsive spectral content. For such signals, the quantization noise in the FDLP codec appears as frequency components not present in the input signal. In this paper, we propose a technique of Spectral Noise Shaping (SNS) for improving the quality of tonal signals by applying a Time Domain Linear Prediction (TDLP) filter prior to the FDLP processing. The inverse TDLP filter at the decoder shapes the quantization noise to reduce the artifacts. Application of the SNS technique to the FDLP codec improves the quality of the tonal signals without affecting the bit-rate. Performance evaluation is done with Perceptual Evaluation of audio Quality (PEAQ) scores and with subjective listening tests.

关键词： Spectral Noise Shaping (SNS) Frequency Domain Linear Prediction (FDLP) speech and audio coding

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PyAWNeS-Codec: speech and audio codec for ad-hoc acoustic wireless sensor networks 29

PyAWNeS-Codec: Speech and audio codec for ad-hoc acoustic wi...

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29th European Signal Processing Conference (EUSIPCO)

作者： Backstrom, Tom Mansali, Mariem Bouafif Zarazaga, Pablo Perez Ranjit, Meghna Das, Sneha Lachiri, Zied Aalto Univ Dept Signal Proc & Acoust Espoo Finland El Manar Univ SITI Lab Natl Engn Sch Tunis Tunis Tunisia

ISBN: (纸本)9789082797060

Existing hardware with microphones can potentially be used as sensor networks to capture speech and audio signals for the benefit of better signal quality than possible with a single microphone. A central pre-requisite for such ad-hoc acoustic wireless sensor networks (ASWNs) is an efficient communication protocol with which to transmit audio data between nodes. For that purpose, we present the world's-first speech and audio codec especially designed for ASWNs, which has competitive quality also in single-channel operation. To ensure quality in the single-channel scenario, it closely resembles conventional codecs of the TCX-type, but extended with features to facilitate multi-device operation, including dithered quantization, delay estimation and compensation, as well as multi-channel post-filtering. The codec is intended to become a baseline for future research and we therefore provide it as an open-access library. Our experiments confirm that performance is in the same range as recent commercial single-channel codecs and that added devices improve quality.

关键词： speech and audio coding ad-hoc acoustic sensor networks time difference of arrival estimation delay compensation multi-channel post-filtering

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Reconstructing audio Signals from Modified Non-Coherent Hilbert Envelopes

Reconstructing Audio Signals from Modified Non-Coherent Hilb...

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Interspeech Conference 2007

作者： Thiemann, Joachim Kabal, Peter McGill Univ Dept Elect & Comp Engn Montreal PQ Canada

ISBN: (纸本)9781605603162

In this paper, we present a speech and audio analysis-synthesis method based on a Basilar Membrane (BM) model. The audio signal is represented in this method by the Hilbert envelopes of the responses to complex gammatone filters uniformally spaced on a critical band scale. We show that for speech and audio signals, a perceptually equivalent signal can be reconstructed from the envelopes alone by an iterative procedure that estimates the associated carrier for the envelopes. The rate requirement of the envelope information is reduced by low-pass filtering and sampling, and it is shown that it is possible to recover a signal without audible distortion from the sampled envelopes. This may lead to improved perceptual coding methods.

关键词： speech and audio coding auditory representation auditory model inversion

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speech quality evaluation of neural audio codecs 25

Speech quality evaluation of neural audio codecs

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25th Interspeech Conference

作者： Muller, Thomas Ragoti, Stephane Gros, Laetitia Philippe, Pierrick Scalart, Pascal Orange Innovat Ile De France France Univ Rennes IRISA Rennes France

This paper presents speech quality results to characterize the state of the art and technological advance of recent neural audio codecs targeting low bitrates. audio quality was evaluated in one clean speech experiment (in French). Degradation Mean Opinion Score (DMOS) results are reported and discussed for neural audio codecs (LPCNet, Lyra V2, EnCodec, audioCraft, audioDec, Descript audio Codec) - traditional codecs (Opus, EVS) are also included as performance yardsticks. We also discuss observed codec complexity to complement subjective test results.

关键词： speech and audio coding subjective test neural audio coding

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AN INTER-FRAME CORRELATION BASED ERROR CONCEALMENT OF IMMITTANCE SPECTRAL COEFFICIENTS FOR MOBILE speech AND audio CODECS 16

AN INTER-FRAME CORRELATION BASED ERROR CONCEALMENT OF IMMITT...

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16th IEEE International Conference on High Performance Computing and Communications HPCC 2014\11th IEEE International Conference on Embedded Software and Systems ICESS 2014\6th International Symposium on Cyberspace Safety and Security CSS 2014

作者： Yang, Yuhong Dong, Shaolong Hu, Ruimin Wang, Yanye Gao, Li Zhang, Maosheng Wuhan Univ Sch Comp Sci Natl Engn Res Ctr Multimedia Software Wuhan Peoples R China

ISBN: (纸本)9781479961238

This paper proposes an inter-frame correlation based error concealment approach for hybrid CELP (Code Excited Linear Prediction) and transform codecs with both good speech and audio quality at moderate bit rates. The proposed scheme is designed to overcome the main challenge due to the diversified characteristics of input signals. The underlying idea is to employ the inter-frame correlation of previous neighborhood frames to circumvent the pitfalls of referring to the unrelated frames, and to enable effective prediction of ISF (Immittance Spectral Frequencies) spectrum coefficients of missing frames from the immediate relative history using linear regression approach. Objective and subjective evaluation results for the proposed approach, in comparison with existing technique of AVS-P10 (audio Video coding of China Standard Part 10 - Mobile speech and audio Codec), provide strong evidence for gains across a variety of speech and audio signals.

关键词： error concealment speech and audio coding Immittance Spectral Frequencies linear regression

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Low-complexity and robust coding mode decision in the EVS coder

Low-complexity and robust coding mode decision in the EVS co...

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IEEE International Conference on Acoustics, speech and Signal Processing

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

ISBN: (纸本)9781467369985

Several state-of-the-art switched audio codecs employ the closed-loop mode decision to select the best coding mode at every frame. The closed-loop mode selection is known to have good performance but also high complexity. The new approach we propose in this paper is a low-complexity version of the closed-loop approach, based on similar decisions which compute the coding distortion of each mode and select the one with the lowest distortion. Our approach differs mainly in the way the coding distortions are calculated. We are able to notably reduce the complexity by only estimating the distortions without encoding and decoding the input for each mode. The new approach was implemented in the EVS codec standard and evaluated both objectively and subjectively. Compared to the closed-loop approach, it yields similar performance and lower complexity.

关键词： speech and audio coding closed-loop mode decision mode selection open-loop switched coding coding mode CLOSED LOOP Electric vehicles OPEN-LOOP codecs ENCODER

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PARAMETRIC BINAURAL audio coding

PARAMETRIC BINAURAL AUDIO CODING

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IEEE International Conference on Acoustics, speech, and Signal Processing

作者： Pasi Ojala Mikko Tammi Miikka Vilermo Nokia Research Center Tampere Finland

ISBN: (纸本)9781424442959;9781424442966

A spatial audio scene consists of discrete audio sources and ambience. The 3D audio image is observed due to the directional sounds, but even more important is the reverberation and so called room effect caused by the properties of the space, the sources and listener location. It is obvious that a human being is able to capture the 3D image using the signal from left and right ear. Hence, two audio channels are sufficient to represent the spatial audio image for the listener. An efficient transmission and representation of a spatial audio image using two channels requires a specific coding and rendering algorithm for the audio content. In this paper we present novel mechanisms to efficiently parameterize, quantize and represent the spatial audio signal.

关键词： Binaural audio speech and audio coding audio Three-Dimensional Images audio signals Reverberation Listeners audio channels

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