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2004
As the latest extension of MPEG-4 Audio coding, MPEG-4 Lossless Audio Coding includes a scalable audio coding solution (SLS) that integrates the functionalities of lossless audio coding, perceptual audio coding, and fine granular scalable audio coding into a single coder framework while providing backward compatibility to MPEG Advanced Audio Coding (AAC) at the bit-stream level. Despite its abundant functionalities, SLS still achieves a compression performance that is comparable to state-of-the-art non-scalable lossless audio coding algorithms. As a result, SLS provides a universal digital audio format for a variety of application domains including professional audio, Internet music, consumer electronics, broadcasting and others. This paper presents the structure of SLS and its latest developments during the MPEG standardization process.
Journal of the Audio …, 2007
Recently the MPEG Audio standardization group has successfully concluded the standardization process on technology for lossless coding of audio signals. A summary of the scalable lossless coding (SLS) technology as one of the results of this standardization work is given. MPEG-4 scalable lossless coding provides a fine-grain scalable lossless extension of the well-known MPEG-4 AAC perceptual audio coder up to fully lossless reconstruction at word lengths and sampling rates typically used for high-resolution audio. The underlying innovative technology is described in detail and its performance is characterized for lossless and near lossless representation, both in conjunction with an AAC coder and as a stand-alone compression engine. A number of application scenarios for the new technology are discussed.
2003
This paper presents a scalable lossless enhancement of MPEG-4 Advanced Audio Coding (AAC). Scalability is achieved in the frequency domain using the Integer Modified Discrete Cosine Transform (IntMDCT), which is an integer approximation of the MDCT providing perfect reconstruction. With this transform, and only minor extension of the bitstream syntax, the MPEG-4 AAC Scalable codec can be extended to a lossless operation. The system provides bit-exact reconstruction of the input signal independent of the implementation accuracy of the AAC core coder. Furthermore, scalability in sampling rate and reconstruction word length is supported.
2004 IEEE International Conference on Acoustics, Speech, and Signal Processing, 2004
Lossless coding will become the latest extension of the MPEG-4 audio standard. In response to a call for proposals, many companies have submitted lossless audio codecs for evaluation. The codec of the Technical University of Berlin was chosen as reference model for MPEG-4 Audio Lossless Coding (ALS), attaining working draft status in July 2003. The encoder is based on linear prediction, which enables high compression even with moderate complexity, while the corresponding decoder is straightforward. The paper describes the basic elements of the codec, points out envisaged applications, and gives an outline of the standardization process.
2012
MPEG-4 Audio Lossless Coding (ALS) is a new extension of the MPEG-4 audio coding family. The ALS core codec is based on forward-adaptive linear prediction, which offers remarkable compression together with low complexity. Additional features include long-term prediction, multichannel coding, and compression of floating-point audio material. In this paper authors who have actively contributed to the standard describe the basic elements of the ALS codec with a focus on prediction, entropy coding, and related tools. We also present latest developments in the standardization process and point out the most important applications of this new lossless audio format.
IEEE Transactions on Consumer Electronics, 2004
MPEG-AAC is the current state of the art in audio compression technology. The CD-quality promised at bit rate as low as 64 kbps makes AAC a strong candidate for high quality low bandwidth audio streaming applications over wireless network. Besides this low bit rate requirement, the codec must be able to run on personal wireless handheld devices with its inherent low power characteristics. While the AAC standard is definite enough to ensure that a valid AAC stream is correctly decodable by all AAC decoders, it is flexible enough to accommodate variations in implementation, suited to different resources available and application areas. This paper reviews various implementation techniques of the encoder. We then proposed our method of an optimized software implementation of MPEG-AAC (LC profile). The coder is able to perform encoding task using half the processing power compared to standard implementation without significant degradation in quality as shown by both subjective listening test and an ITU-R compliant quality testing program (OPERA).
IEEE Transactions on Consumer Electronics, 2003
We present a flexible audio coding system for use in the continuous transmission of high-quality ) stereo-audio data streams, either live or recorded. The system provides both lossless and variable-level lossy quality; quality is selectable to suit a full range of wide-and narrow-band IP networks. As input signals for transmission at the server PC, less than nine PCM sound files in different formats are simultaneously encodable, and the efficiency of simultaneous compression is very high. The system is realized by software that runs on a typical PC. This gives everyone on an IP network the ability to transmit high-quality sound anywhere within the network. An MPEG-4 audio codec provides the core of the lossy coding module.
EURASIP Journal on Audio, Speech, and Music Processing, 2009
In 2003 and 2004, the ISO/IEC MPEG standardization committee added two amendments to their MPEG-4 audio coding standard. These amendments concern parametric coding techniques and encompass Spectral Band Replication (SBR), Sinusoidal Coding (SSC), and Parametric Stereo (PS). In this paper, we will give an overview of the basic ideas behind these techniques and references to more detailed information. Furthermore, the results of listening tests as performed during the final stages of the MPEG-4 standardization process are presented in order to illustrate the performance of these techniques.
2003
This papers presents an embedded fine grain scalable perceptual and lossless audio coding scheme. The enabling technology for this combined perceptual and lossless audio coding approach is the Integer Modified Discrete Cosine Transform (IntMDCT), which is an integer approximation of the MDCT based on the lifting scheme. It maintains the perfect reconstruction property and therefore enables efficient lossless coding in the frequency domain. The close approximation of the MDCT also allows to build a perceptual coding scheme based on the IntMDCT. In this paper a bitsliced arithmetic coding technique is applied to the IntMDCT values. Together with the encoded shape of the masking threshold a perceptually hierarchical bitstream is obtained, containing several stages of perceptual quality and extending to lossless operation when transmitted completely. A concept of encoding subslices is presented in order to obtain a fine adaptation to the masking threshold especially in the range of perceptually transparent quality.
IEEE Signal Processing Magazine, 2001
audio cooling, lossless compression, Internet audio
This article explains the technologies and applications of lossless audio coding. NTT started research on lossless coding of audio signals and proposed the initiation of international standardization in 2002, aiming at high-quality services suitable for broadband networks. As a result of cooperative work with other organizations, the specifications of this technology were officially incorporated in the ISO/IEC MPEG standard published
2004
Lossless coding will become the latest extension of the MPEG-4 audio standard. The lossless audio codec of the Technical University of Berlin was chosen as reference model for MPEG-4 Audio Lossless Coding (ALS). The MPEG-4 ALS encoder is based on linear prediction, which enables high compression even with moderate complexity, while the corresponding decoder is straightforward. The paper describes the
2006
The MPEG-4 Scalable to Lossless (SLS) audio coding is recently being developed to provide a unified solution for highcompression perceptual audio coding and high-quality lossless audio coding. SLS provides efficient Fine Granular Scalable (FGS) coding from AAC core layer to lossless, and achieves reasonable perceptual quality at its scalable coding range using a sequential bit-plane scanning method, which minimizes the audio distortion according to the spectral shape of the core layer quantization errors. In this paper, it is shown that the perceptual quality performance of SLS at intermediate rates can be further improved by incorporating psychoacoustic model into the bit-plane coding process. In addition, it is also found that such an improvement can be achieved by slightly tweaking the original bit-plane coding process of SLS and hence preserving its nice features such as compatibility to lossless coding and low complexity.
This paper discusses the design and implementation of a scalable audio compression scheme that scales up from lossy to lossless compression. Scalable audio compression has been of interest in the audio compression community for some time, with the most obvious attempt at obtaining a solution coming in the form of the MPEG-4 standard [1]. At the same time the increase in bit rates in both mobile communications [2] and the internet's broadband technology means that audio compression algorithms with higher bit rates than currently used, such as MPEG's mp3 [1], can be employed to obtain higher quality. However, the new increased data rates are not necessarily constant, this is especially the case when considering the internet. As such, scalable schemes that can scale to lossless compression have become rather interesting from an application point of view. The scheme presented in this paper achieves lossless compression that is comparable with the state of the art whilst maintain...
2008 IEEE International Conference on Acoustics, Speech and Signal Processing, 2008
A fully scalable audio coding structure based on a novel combination of the non-core MPEG-4 scalable lossless audio coding (SLS), the state-of-the-art psychoacoustic model, joint stereo coding and the perceptually prioritized bit-plane coding is presented in this paper. The psychoacoustic information is implicitly embedded in the scalable bitstream with negligible amount of side information and trivial modi cation to the standardized SLS decoder. Results of extensive evaluation show that the subjective quality of scalable audio is improved signi cantly.
IEE Proceedings - Vision, Image, and Signal Processing, 2006
Current popular Internet audio streaming solutions impose a division between source coding (provided, for example, by MPEG layer III) and channel coding, typically accomplished in the server by means of packet re-transmission. A novel joint source and channel coder that provides packet-loss recovery and continuous bitrate scalability is presented. These functionalities are well suited to streaming audio over third and future generation wireless broadband networks.
Electronics Letters, 2007
A simple and flexible bit-plane coding method is developed for scalable audio coding. It is different from the traditional bit-plane coding in that the optimal bit-plane scanning order is adapted to the scale between the energy of the residual signal and the audio mask. Both the perceptual quality and the lossless compression ratio at common lossy bitrates are greatly improved compared with the performance of state-of-the-art scalable audio.
JOURNAL-AUDIO …, 2007
An overview of the recently finalized ISO/MPEG standard for multichannel audio com-pression MPEG Surround is provided. This audio compression scheme enables backward-compatible multichannel audio coding and transmission at unsurpassed coding efficiency. This is ...
Journal of The Audio Engineering Society, 2005
Audio Lossless Coding (ALS) is a new addition to the suite of MPEG-4 audio coding standards. The ALS codec is based on forward-adaptive linear prediction, which offers remarkable compression even with low predictor orders. Nevertheless, performance can be significantly improved by using higher predictor orders, more efficient quantization and encoding of the predictor coefficients, and adaptive block length switching. The paper describes the basic elements of the ALS codec with a focus on these recent improvements. It also presents the latest developments in the standardization process and describes several important applications of this new lossless audio format in practice.
eurasip.org
Audio compression has progressively gained higher importance in the Internet thanks to massive amount of multimedia services on it. This new services require coders adapted to that new environment. Therefore, new generation coders use more complex models focused on features which make possible its use for audio streaming over the Internet, mainly low bit rate, scalability and robustness. In our case, a good trade-off between bit rate reduction and audio quality is achieved by using parametric audio coding, and furthermore, this coder has a scalable version, optimized for streaming requirements. This coder avoids differential information between coded audio segments and uses a layered scheme for changing straightforwardly the bit rate. The results reveal our coder as a good candidate for massive distributed audio applications, like music on demand, radio broadcasting or real-time streaming audio. In this article are shown the main features of this coder and their implication on streaming.
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