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perceptual coding of narrow-band audio signals at low rates

IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2006年第2期14卷 609-622页

作者： Najaf-Zadeh, H Kabal, P Adv Audio Syst Commun Res Ctr Ottawa ON K2H 8S2 Canada McGill Univ Montreal PQ H3A 2A7 Canada

This paper describes a coding paradigm using coding tools based on the characteristics of the human hearing system so as to accommodate a wide range of narrow-band audio inputs without annoying artifacts at low rates (down to 8 kb/s). The narrow-band perceptual audio coder (NPAC) employs a variety of algorithms to account for the perceptually irrelevant parts of the input signal in addition to statistical redundancies. The new algorithms used in the NPAC coder include a perceptual error measure in training the codebooks and selecting the best codewords which takes into account the audible parts of the quantization noise, a perception-based bit-allocation algorithm and a new predictive scheme to vector quantize the scale factors. The NPAC coder delivers acceptable quality without annoying artifacts for most narrow-band audio signals at around 1 bit/sample. Informal subjective tests have shown that the NPAC coder outperforms a commercial low-rate music coder operating at 8 kb/s.

关键词： adaptive bit allocation masking model modified discrete cosine transform filter bank narrow-band audio coding perceptual audio coding perceptual distortion perceptual vector quantization predictive vector quantization

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Real-Time perceptual Moving-Horizon Multiple-Description audio coding

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IEEE TRANSACTIONS ON SIGNAL PROCESSING 2011年第9期59卷 4286-4299页

作者： Ostergaard, Jan Quevedo, Daniel E. Jensen, Jesper Aalborg Univ Dept Elect Syst DK-9220 Aalborg Denmark Univ Newcastle Sch Elect Engn & Comp Sci Callaghan NSW 2308 Australia Oticon DK-9220 Copenhagen Denmark

A novel scheme for perceptual coding of audio for robust and real-time communication is designed and analyzed. As an alternative to PCM, DPCM, and more general noise-shaping converters, we propose to use psychoacoustically optimized noise-shaping quantizers based on the moving-horizon principle. In moving-horizon quantization, a few samples look-ahead is allowed at the encoder, which makes it possible to better shape the quantization noise and thereby reduce the resulting distortion over what is possible with conventional noise-shaping techniques. It is first shown that significant gains over linear PCM can be obtained without introducing a delay and without requiring postprocessing at the decoder, i.e., the encoded samples can be stored as, e. g., 16-bit linear PCM on CD-ROMs, and played out on standards-compliant CD players. We then show that multiple-description coding can be combined with moving-horizon quantization in order to combat possible erasures on the wireless link without introducing additional delays.

关键词： Low delay source coding moving horizon quantization multiple-description coding perceptual audio coding

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WAVEFORM APPROXIMATING RESIDUAL audio coding WITH perceptual PRE- AND POST-FILTERING

WAVEFORM APPROXIMATING RESIDUAL AUDIO CODING WITH PERCEPTUAL...

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42nd Asilomar Conference on Signals, Systems and Computers

作者： Nielsen, Jesper Kjoer Jensen, Jesper Rindom Christensen, Mads Groesboll Jensen, Soren Holdt Larsen, Torben Aalborg Univ Dept Elect Syst DK-9220 Aalborg Denmark

ISBN: (纸本)9781424429400

We investigate waveform approximating residual coding for a sinusoidal parametric audio coder at low bit rates. The residual coding is based on the well-known pre- and post-filtering method with loss-less coding [1] which features perceptual weighting for short time segments. We compare the incurred perceptual distortion from joint quantization of the residual and the sinusoids for different bit rates. In addition to that, we develop a transform coding scheme for the coefficients in the pre- and post-filters which must be send as side information between the encoder and decoder. Our investigations show that the combination of the sinusoidal subcoder and the pre- and post-filtering entails an overall lower perceptual distortion for low as well as high bit rates. Also, the developed transform coding scheme enables efficient coding of the side information at a low bit rate.

关键词： perceptual audio coding residual coding pre- and post-filtering sinusoidal audio coding

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A new quantization optimization algorithm for the MPEG advanced audio coder using a statistical subband model of the quantization noise

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IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2006年第4期14卷 1328-1339页

作者： Derrien, O Duhamel, P Charbit, M Richard, G ENST TSI Signal & Image Proc Dept F-75634 Paris 13 France Ecole Super Elect LSS CNRS Signals & Syst Lab F-91192 Gif Sur Yvette France

In this paper, an improvement of the quantization optimization algorithm for the MPEG-Advanced audio Coder (AAC) is presented. This algorithm, given a bit-rate constraint, minimizes the perceived distortion generated by the signal compression. The distortion can be related to the quantization error level over frequency subbands through an auditory model. Thus, optimizing the quantization requires knowledge of the rate-distortion function for each subband. When this function can be modeled in a simple way, the algorithm can take a one-loop recursive structure. However, in the MPEG AAC, the rate-distortion function is hard to characterize, since AAC makes use of nonlinear quantizers and variable length entropy coders. As a result, the standard algorithm makes use of two nested loops with a local decoder, in order to measure the error level rather than predicting its value. We first describe a partial subband modeling of the rate-distortion function of interest in the MPEG AAC. Then, using a statistical approach, we find a relationship between the error level and the so-called quantization "scale-factor" and propose a new algorithm that is basically similar to a classical one loop "bit allocation" process. Finally, we describe the complete algorithm and show that it is more efficient than the standard one.

关键词： bit-rate constraint distortion constraint optimization algorithm perceptual audio coding scale-factor statistical model subband quantization

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MDCTNet: A Hybrid Approach to Neural audio coding

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IEEE JOURNAL OF SELECTED TOPICS IN SIGNAL PROCESSING 2024年第8期18卷 1506-1516页

作者： Villemoes, Lars Vinton, Mark Ekstrand, Per Lu, Lie Davidson, Grant Zhou, Cong Dolby Sweden AB Adv Technol Grp S-11330 Stockholm SE Sweden Dolby Labs Inc Adv Technol Grp San Francisco CA 94103 USA Anuttacon Santa Clara CA 95054 USA

We describe and evaluate a hybrid neural audio coding system consisting of a perceptual audio encoder and a generative model, MDCTNet. By applying recurrent layers (RNNs) we capture correlations in both time and frequency directions in a perceptually weighted adaptive modified discrete cosine transform (MDCT) domain. By training MDCTNet on a diverse set of full-range monophonic audio signals at 48 kHz sampling, we achieve performance competitive with state-of-the-art audio coding at 24 kb/s variable bitrate (VBR). We also quantify the effect of the generative model-based decoding at lower and higher bitrates and discuss some caveats of the use of data driven signal reconstruction for the audio coding task.

关键词： Decoding Psychoacoustic models Transforms Codecs Bit rate audio coding Entropy Distortion Training Time-frequency analysis perceptual audio coding deep learning generative models neural networks

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A fine granular scalable to lossless audio coder

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IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2006年第4期14卷 1352-1363页

作者： Yu, Rongshan Rahardja, Susanto Xiao, Lin Ko, Chi Chung Inst Infocomm Res Singapore 119613 Singapore Natl Univ Singapore Dept Elect & Comp Engn Singapore 119620 Singapore

This paper presents Advanced audio Zip (AAZ), a fine grained scalable to lossless (SLS) audio coder that has recently been adopted as the reference model for MPEG-4 audio SLS work. AAZ integrates the functionalities;of high-compression perceptual audio coding, line granular scalable audio coding, and lossless audio coding in a single framework, and simultaneously provides backward compatibility to MPEG-4 Advanced audio coding (AAC). AAZ provides the fine granular bit-rate scalability from lossy to lossless coding, and such a scalability is achieved in a perceptually meaningful way, i.e., better perceptual quality at higher bit-rates. Despite its abundant functionalities, AAZ only introduces negligible overhead in terms of lossless compression performance compared with a nonscalable, lossless only audio coder. As a result, AAZ provides a universal yet efficient solution for digital audio applications such as audio archiving, network audio streaming, portable audio playing, and music downloading which were previously catered for by several different audio coding technologies, and eliminates the need for any transcoding system to facilitate sharing of digital audio contents across these application domains.

关键词： audio compression lossless audio coding perceptual audio coding scalable audio coding

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Toward bandwidth-efficient and error-robust audio streaming over lossy packet networks

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MULTIMEDIA SYSTEMS 2005年第5期10卷 402-412页

作者： Korhonen, J Wang, Y Isherwood, D Nokia Res Ctr Tampere Finland Natl Univ Singapore Singapore 117548 Singapore Nokia Electr Ltd Tampere Finland

Bandwidth efficiency and error robustness are essential issues for different multimedia streaming applications. This paper presents strategies for high-quality audio streaming based on fragmenting perceptually coded audio frames and shuffling the data components among multiple packets for transportation. This is done to increase robustness against packet loss. We also address the delivery of audio data consisting of components with different proportional priorities. Our approach is rationalized with streaming tests using the MPEG AAC audio codec in a simulated network environment and formal listening tests to evaluate the resulting audio output. According to the results, the proposed schemes improve audio quality significantly with reasonable increase to network resource utilization compared to traditional error robustness measures.

关键词： audio streaming multimedia networking perceptual audio coding teleconferencing advanced audio coding (AAC)

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A New Model-Based Algorithm for Optimizing the MPEG-AAC in MS-Stereo

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IEEE TRANSACTIONS ON audio SPEECH AND LANGUAGE PROCESSING 2008年第8期16卷 1373-1382页

作者： Derrien, Olivier Richard, Gael Univ Sud Toulon Var ISITV F-83957 La Garde France CNRS LTCI Inst TELECOM F-75014 Paris France

In this paper, a new model-based algorithm for optimizing the MPEG-Advanced audio Coder (AAC) in MS-stereo mode is presented. This algorithm is an extension to stereo signals of prior work on a statistical model of quantization noise. Traditionally, MS-stereo coding approaches replace the Left (L) and Right (R) channels by the Middle (M) and Sides (S) channels, each channel being independently processed, almost like a monophonic signal. In contrast, our method proposes a global approach for coding both channels in the same process. A model for the quantization error allows us to tune the quantizers on channels M and S with respect to a distortion constraint on the reconstructed,channels L and R as they will appear in the decoder. This approach leads to a more efficient perceptual noise-shaping and avoids using complex psychoacoustic models built on the M and S channels. Furthermore, it provides a straightforward scheme to choose between LR and MS modes in each subband for each frame. Subjective listening tests prove that the coding efficiency at a medium bitrate (96 kbits/s for both channels) is significantly better with our algorithm than with the standard algorithm, without increase of complexity.

关键词： Bitrate constraint distortion constraint MPEG-Advanced audio Coder (AAC) MS-stereo optimization algorithm perceptual audio coding scalefactor statistical model quantization

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Frame splitting scheme for error-robust audio streaming over packet-switching networks

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IEICE TRANSACTIONS ON COMMUNICATIONS 2008年第2期E91B卷 677-680页

作者： Kim, Jong Kyu Kim, Jung Su Yun, Hwan Sik Chang, Joon-Hyuk Kim, Nam Soo Seoul Natl Univ Sch Elect Engn Seoul 151744 South Korea Seoul Natl Univ INMC Seoul 151744 South Korea Inha Univ Sch Elect & Elect Engn Inchon 401751 South Korea

This letter presents a novel frame splitting scheme for an error-robust audio streaming over packet-switching networks. In our approach to perceptual audio coding, an audio frame is split into several sub-frames based on the network configuration such that each packet can be decoded independently at the receiver. Through a subjective comparison category rating (CCR) test, it is discovered that our approach enhances the quality of the decoded audio signal under,the lossy packet-switching networks environment.

关键词： perceptual audio coding audio streaming frame splitting packet-switching networks

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On the use of backward adaptation in a perceptual audio coder

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IEEE TRANSACTIONS ON SPEECH AND audio PROCESSING 2000年第4期8卷 488-490页

作者： Rodrigues, JM Tomé, AM Univ Aveiro Dept Elect & Telecommun IEETA P-3810 Aveiro Portugal

Typically, perceptual audio coders have followed a subband or transform coding scheme with forward-adaptive quantization,In this letter we present an alternative scheme which uses backward-adaptive quantization, We di... 详细信息

关键词： adaptive coding perceptual audio coding

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