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MPEG-4 Natural Audio Features
Just as in MPEG-1 and MPEG-, Audio compression is the core functionality of MPEG-4. However, the range of bit rates covered by MPEG-4 Audio and the compression ratio and sound quality have been improved much over previous systems and can be considered state of the art today. Besides this MPEG-4 has a lot additional new functionalities. Fraunhofer IIS provides technology supporting the following functions
•Bit rate scalability (hierarchical coding) allows a bit stream to be parsed into bit streams of lower bit rate which still can be decoded into a meaningful signal. The bit stream parsing can occur either during transmission or in the decoder.
•Bandwidth scalability is a particular case of bit rate scalability, whereby part of a bit stream representing a part of the frequency spectrum can be discarded during transmission or decoding.
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•Encoder complexity scalability allows encoders of different complexity to generate valid and meaningful bit streams.
•Decoder complexity scalability allows a given bit stream to be decoded by decoders of different levels of complexity. The audio quality, in general, is related to the complexity of the encoder and decoder used.
•Error robustness tools provide improved performance on error-prone transmission channels. These tools consist of codec specific error resilience tools and a common error protection tool.
•Low Delay Audio Coding allows for high quality coding of general audio signals with a delay of just around 0ms (important for real-time bi-directional communications).
The MPEG-4 standard is designed to play the role of a global multimedia language. Within its source coding parts, MPEG-4 embraces a multitude of natural and synthetic audio and video coding schemes and representations, providing a generic tool set for numerous applications for the transmission and storage of audiovisual signals. In addition, MPEG-4 introduces and supports completely new concepts of object-based user interactivity as well as a rich feature set to maintain a useful quality of service within error-prone channels.
Coding scheme
The basis of MPEG-4 video coding is a block-based predictive differential video coding scheme. The main techniques for compression are
•Division of the picture in 8x8-blocks or 16x16-macroblocks (MB)
•Motion compensated prediction
•Transform coding with discrete cosine transform (DCT)
•Quantization
•Run length and huffman coding for variable length codes (VLC)
There are two basic coding modes available
Intra-Mode
Both the spatial redundancy and irrelevancy are exploited with block based DCT coding, quantization, run length and huffman coding. Only information from the picture itself is used and thus every frame can be decoded independently.
Inter-Mode
Additionally, the temporal redundancy between the pictures in a video sequence is taken into account. A macroblock based motion estimation between two successive images is done allowing a motion compensated prediction of the current picture. Afterwards the predicted image is subtracted from the original image. The resulting difference picture is DCT coded, quantized and VLC coded. The motion vectors describing the motion of the blocks in the picture are necessary side information for the decoder and are also encoded with VLC.
Interframe coding usually requires much lower bit rate in comparison to intraframe coding (only about 10 to 0%) because there is less information in the difference picture than in the original picture.
MPEG-4 offers extensive systems part supporting scene descriptors, object descriptors and all related features, such as Intellectual Property Management and Protection (IPMP) provisions. This system allows for simple (old style) single source audio coding as well as complex, object oriented audio scenes. In order to make full use of all of the MPEG-4 features, a transport mechanism must support the MPEG-4 systems requirements.
In addition MPEG-4 defines the following transport formats, although other transmission systems with different application specific transport formats are possible as well
•MPEG-4 Flexmux. This is not really a transport format but rather an interface description between MPEG-4 and an arbitrary transport format. During the development of MPEG-4 it served with some additions as a simple file format.
•MPEG-4 File Format. This is the file format for MPEG-4 content. It is derived from Apples Quicktime™ media format and supports e.g. editing and streaming of MPEG-4 content.
•LOAS (Low overhead audio stream). This is an audio-only transport format for applications where an MPEG-4 audio object needs to be transmitted and additional transport overhead is an issue.
•LATM (Low overhead Audio Transport Multiplex). This is the multiplex part of the LOAS described above. Can be used if only a multiplex is needed.
•MPEG-4 over RTP (Real Time Protocol). This is protocol defined by the IETF (Internet Engineering Task Force), but there are several ways how to transmit MPEG-4 content over IP networks (e.g. the Internet) using RTP. One possibility to transmit MPEG-4 Audio is to use LATM and RTP.
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