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检索条件"主题词=Audio coding"

共 1197 条记录，以下是81-90 订阅

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HYBRID NOISE SHAPING FOR audio coding USING PERFECTLY OVERLAPPED WINDOW

HYBRID NOISE SHAPING FOR AUDIO CODING USING PERFECTLY OVERLA...

引用

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

作者： Jo, Byeongho Beack, Seungkwon Elect & Telecommun Res Inst Daejeon South Korea

ISBN: (纸本)9798350323726

In recent years, audio coding technology has been standardized based on several frameworks that incorporate linear predictive coding (LPC). However, coding the transient signal using frequencydomain LP residual signals remains a challenge. To address this, temporal noise shaping (TNS) can be adapted, although it cannot be effectively operated since the estimated temporal envelope in the modified discrete cosine transform (MDCT) domain is accompanied by the time-domain aliasing (TDA) terms. In this study, we propose the modulated complex lapped transform-based coding framework integrated with transform coded excitation (TCX) and complex LPC-based TNS (CTNS). Our approach uses a 50% overlap window and switching scheme for the CTNS to improve the coding efficiency. Additionally, an adaptive calculation of the target bits for the sub-bands using the frequency envelope information based on the quantized LPC coefficients is proposed. To minimize the quantization mismatch between both modes, an integrated quantization for real and complex values and a TDA augmentation method that compensates for the artificially generated TDA components during switching operations are proposed. The proposed coding framework shows a superior performance in both objective metrics and subjective listening tests, thereby demonstrating its low bit-rate audio coding.

关键词： audio coding quantization MCLT window function temporal noise shaping

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ADPCM with adaptive pre- and post-filtering for delay-free audio coding

ADPCM with adaptive pre- and post-filtering for delay-free a...

引用

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

作者： Holters, Martin Pabst, Oliver Zoelzer, Udo Helmut Schmidt Univ Univ Fed Armed Forces Dept Signal Proc & Commun Hamburg Germany

ISBN: (纸本)9781424407286

We propose an extension of ADPCM that includes adaptive pre- and post-filtering to achieve spectral shaping of the coding noise. The advantage of this coding scheme is that it allows a realization without algorithmic delay by making the filters backwards-adaptive. The measurements we present indicate that the addition of adaptive pre- and post-filtering to ADPCM results in a significant improvement in perceived audio quality. We therefore believe that the proposed system is a viable way to near-transparent lossy audio coding without algorithmic delay.

关键词： audio coding linear predictive coding adaptive filters

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UNIFIED SPEECH AND audio coding SCHEME FOR HIGH QUALITY AT LOW BITRATES

UNIFIED SPEECH AND AUDIO CODING SCHEME FOR HIGH QUALITY AT L...

引用

IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Neuendorf, M. Gournay, P. Multrus, M. Lecomte, J. Bessette, B. Geiger, R. Bayer, S. Fuchs, G. Hilpert, J. Rettelbach, N. Salami, R. Schuller, G. Lefebvre, R. Grill, B. Fraunhofer IIS Erlangen Germany Univ Sherbrooke Sherbrooke PQ Canada VoiceAge Corp Montreal PQ Canada Fraunhofer IDMT Ilmenau Germany

ISBN: (纸本)9781424423538

Traditionally, speech coding and audio coding were separate worlds. Based on different technical approaches and different assumptions about the source signal, neither of the two coding schemes could efficiently represent both speech and music at low bitrates. This paper presents a unified speech and audio codec, which efficiently combines techniques from both worlds. This results in a codec that exhibits consistently high quality for speech. music and mixed audio content. The paper gives an overview of the codec architecture and presents results of formal listening tests comparing this new codec with HE-AAC(v2) and AMR-WB+. This new codec forms the basis of the reference model in the ongoing MPEG standardization activity for Unified Speech and audio coding.

关键词： audio coding speech coding

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End-to-end Stereo audio coding Using Deep Neural Networks

End-to-end Stereo Audio Coding Using Deep Neural Networks

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14th Annual Summit and Conference of the Asia-Pacific-Signal-and-Information-Processing-Association (APSIPA ASC)

作者： Lim, Wootaek Jang, Inseon Beack, Seungkwon Sung, Jongmo Lee, Taejin Elect & Telecommun Res Inst Daejeon South Korea

ISBN: (数字)9786165904773

ISBN: (纸本)9786165904773

Recently, deep neural-network-based audio data compression has been widely studied. Although most studies in this area have focused on mono channel speech coding, research on stereo coding is required for ultimate audio coding. In this paper, we consider in case of the stereo channel and present a method of end-to-end stereo audio coding, which exploits the concept of a mid/side (M/S) stereo coding scheme in the latent space of neural network. The goal of this study is to create an end-to-end neural network to find the optimal transformation for stereo coding. Therefore, the latent space conceptually corresponds to M and S, but better feature transformation for stereo coding is learned through the network. Through the end-to-end learning, the proposed method provides a more efficient bit allocation result than the discrete coding of each stereo channel into mono. According to the results of an objective test, the proposed method shows a superior performance in comparison to a discrete stereo model and conventional HE-AAC. A subjective evaluation shows that, for a bitrate of 64 kbps, the proposed model provides a significantly better sound quality than the discrete stereo coding model, as well as a performance comparable to that of a conventional HE-AAC.

关键词： Deep learning Performance evaluation Analytical models audio coding Speech coding Neural networks Bit rate

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Frequency domain linear prediction for QMF sub-bands and applications to audio coding

Frequency domain linear prediction for QMF sub-bands and app...

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4th International Workshop on Machine Learning for Multimodal Interaction

作者： Motlicek, Petr Ganapathy, Sriram Hermansky, Hynek Garudadri, Harinath IDIAP Res Inst Av Pres Beuddin 20 CH-1920 Martigny Switzerland Ecole Polytech Fed Lausanne Lausanne Switzerland Qualcomm Inc San Diego CA USA

ISBN: (纸本)9783540781547

This paper proposes an analysis technique for wide-band audio applications based on the predictability of the temporal evolution of Quadrature Mirror Filter (QMF) sub-band signals. The input audio signal is first decomposed into 64 sub-band signals using QMF decomposition. The temporal envelopes in critically sampled QMF sub-bands are approximated using frequency domain linear prediction applied over relatively long time segments (e.g. 1000 ms). Line Spectral Frequency parameters related to autoregressive models are computed and quantized in each frequency sub-band. The sub-band residuals are quantized in the frequency domain using a combination of split Vector Quantization (VQ) (for magnitudes) and uniform scalar quantization (for phases). In the decoder, the sub-band signal is reconstructed using the quantized residual and the corresponding quantized envelope. Finally, application of inverse QMF reconstructs the audio signal. Even with simple quantization techniques and without any sophisticated modules, the proposed audio coder provides encouraging results in objective quality tests. Also, the proposed coder is easily scalable across a wide range of bit-rates.

关键词： audio coding Frequency Domain Linear Prediction (FDLP) Perceptual Evaluation of audio Quality (PEAQ)

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A MODIFIED DISTORTION METRIC FOR audio coding

A MODIFIED DISTORTION METRIC FOR AUDIO CODING

引用

IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Melkote, Vinay Rose, Kenneth Univ Calif Santa Barbara Dept Elect & Comp Engn Santa Barbara CA 93106 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 perceptual distortion lapped transform modified discrete sine transform

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HIGH-QUALITY audio coding USING MULTIPULSE LPC

HIGH-QUALITY AUDIO CODING USING MULTIPULSE LPC

引用

1990 INTERNATIONAL CONF ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING ( ICASSP 90 )

作者： SINGHAL, S Bell Communications Research Inc. Morristown NJ USA

Experiments are described in coding broadband audio using multipulse linear predictive coding (LPC). It is possible to obtain stable LPC filters that model sinusoids closely and to include perceptual masking in these coders. The quantization of both the LPC and multipulse parameters is also examined, and it is found that multipulse can compensate for quantization error in LPC filters. With appropriate perceptual masking, these coders can provide high quality and audio output. At 128 kb/s, the coders achieved typical SNR values of 35-40 dB in simulations.< >

关键词： audio coding Linear predictive coding Filters Quantization

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HARP-NET: HYPER-AUTOENCODED RECONSTRUCTION PROPAGATION FOR SCALABLE NEURAL audio coding

HARP-NET: HYPER-AUTOENCODED RECONSTRUCTION PROPAGATION FOR S...

引用

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

作者： Petermann, Darius Beack, Seungkwon Kim, Minje Indiana Univ Dept Intelligent Syst Engn Bloomington IN 47408 USA Elect & Telecommun Res Inst Daejeon 34129 South Korea

ISBN: (纸本)9781665448703

We propose a novel autoencoder architecture that improves the architectural scalability of general-purpose neural audio coding models. An autoencoder-based codec employs quantization to turn its bottleneck layer activation into bitstrings, a process that hinders information flow between the encoder and decoder parts. To circumvent this issue, we employ additional skip connections between the corresponding pair of encoder-decoder layers. The assumption is that, in a mirrored autoencoder topology, a decoder layer reconstructs the intermediate feature representation of its corresponding encoder layer. Hence, any additional information directly propagated from the corresponding encoder layer helps the reconstruction. We implement this kind of skip connections in the form of additional autoencoders, each of which is a small codec that compresses the massive data transfer between the paired encoder-decoder layers. We empirically verify that the proposed hyperautoencoded architecture improves perceptual audio quality compared to an ordinary autoencoder baseline.

关键词： audio coding deep learning U-Net autoencoders

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Robust signal/noise discrimination for wideband speech and audio coding

Robust signal/noise discrimination for wideband speech and a...

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7th IEEE Workshop on Speech coding

作者： Jelinek, M Labonté, F Univ Sherbrooke Dept Elect & Comp Engn Sherbrooke PQ J1K 2R1 Canada

ISBN: (纸本)0780364163

We present a robust discrimination method to separate information carrying signals from ambient noise for wideband coding applications. The method consists of a two-stage procedure. First. a local decision is made based on a set of extracted parameters to update the estimated noise level. The parameters has been chosen to detect reliably speech as well as music signals. Then, the final decision is made based only on a frequency dependent Signal to Noise Ratio (SNR). The noise level update does not depend on the final decision to prevent the discriminator from locking when noise level changes suddenly. The performance is compared with the performance of the Voice Activity Detector (VAD) of G.729, Annex G.

关键词： audio coding Multiple signal classification Noise level Noise robustness Signal detection Signal processing Signal to noise ratio Speech coding Speech enhancement Wideband

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Packet loss concealment in predictive audio coding

Packet loss concealment in predictive audio coding

引用

Workshop on Applications of Signal Processing to audio and Acoustics

作者： Wabnik, S Schuller, G Hirschfeld, J Kraemer, U Fraunhofer Inst Digital Media Technol D-98693 Ilmenau Germany

ISBN: (纸本)0780391543

In this paper we present and evaluate Several concealment strategies for packet losses in the context of it low delay predictive audio coder. Our goal is to minimize the audible impact of a packet loss. The problem is that the Predictive coder is backward adaptive, hence depending on past values. There is a predictor reset, but to increase coding efficiency. the distance between two resets is several hundred packets. Hence, not only the lost packet itself cannot be reconstructed, but the transmitted data LIP to the next reset does not result in an exact reconstruction of the audio signal. Our approach is to try to use its much information as possible from the still available data until the next reset. and to reconstruct an audio signal such that distortions are least objectionable. To compare different approaches, we conducted it listening test. The result is that an adaptive reconstruction filter works best in this context.

关键词： audio coding adaptive reconstruction filter audible impact audio signal coding efficiency packet loss concealment packet losses predictive audio coding Packet loss audio coding audio signals code efficiency audio coder Data Security

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