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Parametric audio coding

Parametric audio coding

引用

International Conference on Communication Technology (ICCT)

作者： B. Edler H. Purnhagen Laboratorium für Informationstechnologie University of Hannover Hanover Germany

For very low bit rate audio coding applications in mobile communications or on the Internet, parametric audio coding has evolved as a technique complementing the more traditional approaches. These are transform codecs originally designed for achieving CD-like quality on one hand, and specialized speech codecs on the other. Both of these techniques usually represent the audio signal waveform in such a way that the decoder output signal gives an approximation of the encoder input signal, while taking into account perceptual criteria. Compared to this approach, in parametric audio coding the models of the signal source and of human perception are extended. The source model is now based on the assumption that the audio signal is the sum of "components," each of which can be approximated by a relatively simple signal model with a small number of parameters. The perception model is based on the assumption that the sound of the decoder output signal should be as similar as possible to that of the encoder input signal. Therefore, the approximation of waveforms is no longer necessary. This approach can lead to a very efficient representation. However, a suitable set of models for signal components, a good decomposition, and a good parameter estimation are all vital for achieving maximum audio quality. We give an overview of the current status of parametric audio coding developments and demonstrate the advantages and challenges of this approach. Finally, we indicate possible directions of further improvements.

关键词： audio coding Bit rate MPEG 4 Standard Decoding Speech coding Laboratories Internet Speech codecs Parameter estimation Transform coding

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Error robustness scheme for perceptually coded audio based on interframe shuffling of samples

Error robustness scheme for perceptually coded audio based o...

引用

IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Korhonen, J Nokia Res Ctr Tampere Finland

ISBN: (纸本)0780374029

High-quality audio streaming over IP networks is a significant part of multimedia traffic in telecommunications networks in the future. Although there are a number of error concealment and correction methods developed for real-time multimedia streaming over unreliable packet-switched networks, many effective codec dependent error robustness schemes have not been utilized for the state-of-art audio compression standards developed mainly to compress audio for storage media. This paper describes how the idea of redistributing adjacent audio samples to different packets can be applied to the perceptual audio codecs, such as MP3 and AAC. The experiences with testing the concept using AAC are explained as well. The proposed approach is especially applicable with semi-reliable transport protocols and future networks providing flexible support for prioritized data traffic.

关键词： audio coding codecs data compression multimedia communication telecommunication traffic transport protocols error correction codes code standards signal sampling Internet high-quality audio streaming IP networks multimedia traffic telecommunications networks real-time multimedia streaming error robustness perceptually coded audio interframe sample shuffling codec audio compression standards adjacent audio sample redistribution perceptual audio codecs MP3 AAC semi-reliable transport protocols prioritized data traffic

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Embedding and detecting spread-spectrum watermarks under estimation attacks

Embedding and detecting spread-spectrum watermarks under est...

引用

IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Kirovski, D Malvar, H Microsoft Res Redmond WA 98052 USA

ISBN: (纸本)0780374029

Spread-spectrum (SS) watermarking has been one of the oldest methodologies for hiding data in multimedia signals. Recently, it has been demonstrated that simple block repetition codes provide strong robustness of such watermarks to noise addition and limited arbitrary geometric transformations. In order to further investigate the security of such watermarks, we explore in this paper their robustness with respect to watermark estimation attacks. In such attacks, the adversary has knowledge of the watermark algorithm, except the secret keys. We present a modification of the traditional SS watermark detector that forces the adversary to increase the amount of noise to be proportional to the signal amplitude, in order to remove an SS watermark.

关键词： copy protection data encapsulation audio coding pattern recognition spread-spectrum watermarking SS watermarking multimedia signals block repetition codes noise addition geometric transformations security robustness watermark estimation attacks watermark algorithm adversary signal amplitude

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Psycho-acoustic modeling of audio with exponentially damped sinusoids

Psycho-acoustic modeling of audio with exponentially damped ...

引用

IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Hermus, K Verhelst, W Wambacq, P Katholieke Univ Leuven ESAT Dept Elect Engn PSILab Proc Speech & Images Louvain Belgium

ISBN: (纸本)0780374029

While a traditional sinusoidal model is capable of representing audio segments, a sum of exponentially damped sinusoids is more efficient to model the transient segments that are readily found in audio signals. In this paper, Total Least Squares (TLS) algorithms are applied to automatically extract the modeling parameters in the Exponential Sinusoidal Model (ESM). In order to turn the SNR optimization criterion of these TLS algorithms into a perceptual modeling strategy we incorporate the psycho-acoustic model of MPEG 1-Layer 1 into a subband TLS-ESM scheme. This allows us to model each subband in accordance with its perceptual relevance. Informal listening tests confirm that perceptual ESM achieves the same perceived quality as plain ESM while using substantially less components.

关键词： audio coding hearing transients least squares approximations optimisation psychoacoustic modeling audio signals exponentially damped sinusoids total least squares algorithms exponential sinusoidal model SNR optimization perceptual modeling strategy MPEG-1 Layer 1 audio encoding

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Temporal watermarking invertible only through source separation

Temporal watermarking invertible only through source separat...

引用

IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Berman, G Jouny, I Lafayette Coll Dept Elect & Comp Engn Easton PA 18042 USA

ISBN: (纸本)0780374029

This paper presents an application of blind signal separation to the field of digital signatures and watermarking of audio signals. The second-order blind identification (SOBI) algorithm presented by Belouchrani et al. can be used to retrieve a watermark or to identify a signature. [1]. Multiple source signals (one of which acts as a digital signature) are arbitrarily mixed with grossly different proportions and transmitted. Only the linear mixtures are observed and no information is known about the prior mixing. This process allows for digital verification of the original message, as well as the ability to transmit hidden, buried messages.

关键词： watermarking audio coding copyright blind source separation identification data compression message authentication temporal watermarking source separation blind signal separation digital signatures audio signals watermarking second-order blind identification algorithm watermark retrieval digital signature identification linear mixtures digital verification hidden message transmission buried message transmission audio stream compression decoding copyright protection audio coding

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A new audio coder using a warped linear prediction model and the wavelet transform

A new audio coder using a warped linear prediction model and...

引用

IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Ning, D Deriche, M Queensland Univ Technol Sch Elect Elect & Syst Engn Signal Proc Res Ctr Brisbane Qld 4001 Australia

ISBN: (纸本)0780374029

In this paper, we present results for a wavelet transform (WT) excited warped linear prediction (WLP) audio coder. In contrast to conventional LP, VVLP allows for the control of frequency resolution to closely match the response of the human auditory system. The structure of the system is similar to the transform-coded excitation techniques used in wideband speech coding, where LP has been replaced with WLP. Quantisation of the wavelet coefficients is aided by a psychoacoustic model to minimise the perceptually significant noise due to quantisation error. For monophonic signals sampled at 44.1 kHz, the coder achieves near transparent quality for a variety of speech and music signals at an average bit-rate of 64 kb/s. When compared to MPEG layer III at the same bit-rate, the coder delivers superior quality. The power of the proposed coder resides in its easy scalability to lower bitrates.

关键词： audio coding linear predictive coding transform coding wavelet transforms hearing quantisation (signal) speech coding signal sampling audio coding warped linear prediction wavelet transform frequency resolution human auditory system transform coding psychoacoustic model monophonic signals signal sampling music signals speech coding MPEG layer III scalability 44.1 kHz 64 kbit/s

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Optimal frequency-differential encoding of sinusoidal model parameters

Optimal frequency-differential encoding of sinusoidal model ...

引用

IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Jensen, J Heusdens, R Delft Univ Technol Dept Mediamat Delft Netherlands

ISBN: (纸本)0780374029

Sinusoidal coding has proven to be efficient for low bit-rate audio coding. In this paper we consider schemes for frequency-differential (FD) encoding of the sinusoidal model parameters. For a given signal frame, the parameters of a sinusoidal component may be encoded either differentially relative to other components in the same frame, or directly, i.e., without differential encoding. Using basic tools from graph theory, two algorithms are derived for finding bit rate optimal combinations of direct and differential encoding of the sinusoidal parameters. In simulation experiments with audio signals, the algorithms showed bit-rate reductions of up to 27% relative to direct encoding. Furthermore, when compared to a commonly used FD encoding scheme, the proposed algorithms achieved bit rate reductions of up to 7%.

关键词： optimisation audio coding data compression graph theory optimal frequency-differential encoding sinusoidal model parameters sinusoidal coding graph theory bit rate optimal encoding simulation experiments audio signals bit-rate reduction low bit-rate audio compression low bit-rate audio coding

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Wireless transmission of packet audio using multiple descriptions

Wireless transmission of packet audio using multiple descrip...

引用

13th IEEE International Symposium on Personal, Indoor and Mobile Radio Communications

作者： Barros, J Oikonomidis, I Tech Univ Munich Inst Commun Engn LNT D-80290 Munich Germany

ISBN: (纸本)0780375890

We consider a scenario, where a mobile user requests a high rate audio stream from the internet via a wireless communications system. The congestion on the network and the instability of the wireless channel result in a loss of audio packets, which can impose a severe degradation on the high quality audio signal. To overcome this problem, we propose the use of multiple descriptions (MD). Our novel MD audio codec generates two complementary audio streams, which axe sent in distinct packets to obtain the largest possible diversity gain. Preliminary simulation results support the flexibility and robustness of the proposed solution.

关键词： wireless communications packet networks audio coding multiple descriptions

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Rate-distortion optimal sinusoidal modeling of audio and speech using psychoacoustical matching pursuits

Rate-distortion optimal sinusoidal modeling of audio and spe...

引用

IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Heusdens, R van de Par, S Delft Univ Technol Dept Mediamat NL-2628 CD Delft Netherlands

ISBN: (纸本)0780374029

In this paper, we propose a rate-distortion optimal algorithm for sinusoidal modeling of audio and speech. The algorithm uses a variable-length analysis window where the total number of sinusoids needed to model the source signal is optimally distributed over the segments. To account for human auditory perception, we use a new perceptually relevant distortion measure which is combined with the psychoacousticaI matching pursuit algorithm to select the desired sinusoidal components. We discuss the encoding of the segmentation information and show how to reduce this overhead by restricting the minimum and maximum segment size of the constituent segments. Although this restricts the number of possible partitionings of the input signal, we still have a high accuracy in time at which new segments can start. By doing so, we can decrease the segmentation overhead by 50%, almost without loss of coding efficiency and without introducing pre-echoes.

关键词： audio coding segmentation information human auditory perception psychoacoustical matching pursuits speech. coding sinusoidal modeling rate-distortion optimal algorithm hearing rate distortion theory speech coding

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DSP multi-channel audio decoding

DSP multi-channel audio decoding

引用

IEEE Workshop on Speech coding Proceedings

作者： G. Wineinger Texas Instrumenits Inc. USA

Summary form only given. The multitude of audio coding schemes will continue to increase, as more and diverse ways are found to get closer to real 3D audio. The challenges in keeping up with new formats are best handled by programmable DSPs; however, standardization or pseudstandardization yields to more cost effective approaches. Programmable DSPs give a faster time to market and have lower engineering costs than patching or re-developing a custom VLSI design. Once the decoding algorithm has become an industry standard, custom or ASIC solutions usually yield a more cost effective approach. We discuss some of these approaches and their advantages and disadvantages and the reasons to choose one over the other. Special purpose hardware often consumes much less power, but other factors to be considered are the required interfaces and whether they are likely to change. Deciding which option to pursue and researching and understanding the objectives are also key factors to a successful solution. We must never underestimate the design time, effort, and cost to be incurred in completing the full custom, ASIC development, or programmable DSP. A robust solution is necessary to compete. The issues that surround audio decoding solution alternatives require to be discussed.

关键词： Digital signal processing Decoding Costs Application specific integrated circuits audio coding Standardization Time to market Design engineering Very large scale integration Hardware

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