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The driving force to develop AAC was the quest for an efficient coding method for surround signals, like 5-channel signals (left, right, centre, left-surround, and right-surround) as being used in cinemas today. There have been algorithms for these signals in MPEG- for quite a while. Optimum efficiency, however, was not reached due to technical and historical reasons. Therefore, the set aim was a considerable decrease of necessary bit rate.
MPEG- AAC is the consequent continuation of the truly successful coding method ISO/MPEG Audio Layer- developed in Erlanger. The extensive co-operation with international partners and the insight derived from Layer- paved the way for this coding method, which is unique at this present stage. The appropriate incorporation of high coding gain and great flexibility opens up a wide field of applications. With sampling frequencies between 8 kHz and 6 kHz and any number of channels between 1 and 48, the method is well prepared for future developments in the audio sector. Compared to well-known coding methods such as MPEG- Layer-, it is possible to cut the required bit rate by a factor of two with no loss of subjective quality.
Like all perceptual coding schemes, MPEG- AAC basically makes use of the signal masking properties of the human ear in order to reduce the amount of data. Doing so, the noise is distributed to frequency bands in such a way that it is masked by the total signal, i.e. it remains inaudible. Even though the basic structure of this coding method hardly differs from the ones of its predecessors, a closer look into the details does reveal some new aspects worth paying attention to. The crucial differences between MPEG- AAC and its predecessor ISO/MPEG Audio Layer- are shown as follows
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•Filter bank in contrast to the hybrid filter bank of ISO/MPEG Audio Layer- - chosen for reasons of compatibility but displaying certain structural weaknesses - MPEG- AAC uses a plain Modified Discrete Cosine Transform (MDCT). Together with the increased window length (104 instead of 576 spectral lines per transform) the MDCT outperforms the filter banks of previous coding methods.
•Temporal Noise Shaping (TNS) A true novelty in the area of time/frequency coding schemes. It shapes the distribution of quantization noise in time by prediction in the frequency domain. In particular voice signals experience considerable improvement through TNS.
•Prediction A technique commonly established in the area of speech coding systems. It benefits from the fact that certain types of audio signals are easy to predict.
•Quantization by allowing finer control of quantization resolution, the given bit rate can be used more efficiently.
•Bit-stream format the information to be transmitted undergoes entropy coding in order to keep redundancy as low as possible. The optimization of these coding methods together with a flexible bit-stream structure has made further improvement of the coding efficiency possible.
After MPEG-1 and MPEG-, two standards only concerned about audio and video compression, a new standard called MPEG-4 is fast coming up. MPEG-4 is meant to become the universal language between broadcasting, movie and multimedia applications. It will provide additional functionality over simple media compression, like bit rate scalability, object-based representation, intellectual property management & protection etc., and is based on a rich tool set starting at bit rates as low as kbit/s for a single audio channel.
Scope and Features of the MPEG-4 Standard
The MPEG-4 standard provides a set of technologies to satisfy the needs of authors, service providers and end users alike.
•For authors, MPEG-4 enables the production of content that has far greater reusability and flexibility than is possible with individual technologies, such as digital television, animated graphics, World Wide Web (WWW) pages and their extensions.
•For network service providers, MPEG-4 offers transparent signalling messages suitable for many kinds of networks. For Quality of Service (Quos) considerations, MPEG-4 provides a generic Quos descriptor for different MPEG-4 media, enabling transport optimization in heterogeneous networks.
•For end users, MPEG-4 brings higher levels of interaction with content. It also brings multimedia to new networks, including those employing relatively low bit rate, and mobile ones.
•For all parties involved, MPEG seeks to avoid a multitude of proprietary, non-interworking formats and players.
MPEG-4 achieves these goals by providing consistent ways to
1.Represent units of aural, visual or audiovisual content, called media objects. These media objects can be of natural or synthetic origin; this means they could be recorded with a camera or microphone, or generated with a computer;
.Describe the composition of these objects to create compound media objects that form audiovisual scenes;
.Multiplex and synchronize the data associated with media objects, so that they can be transported over network channels providing a Quos appropriate for the nature of the specific media objects; and
4.Interact with the audiovisual scene generated at the receiver's end.
Errors are everywhere
Whenever digital data is transmitted in real-time, i.e. without the chance of re-transmission, whether in packet oriented networks (e.g. Internet) or in stream-oriented networks (e.g. digital broadcast systems or mobile communication networks), the receiver of this digital data will have to cope with transmission errors. In compressed audio, decoding of corrupted bit streams can lead to annoying artifacts that heavily reduce the audio quality. Those artifacts may even damage the listeners' ears or electronic equipment.
Thus, measures have to be taken to deal with such transmission errors. Four different approaches can be combined
1.Error Detection
adds cyclic redundancy codes to detect errors.
.Error Concealment
synthesizes lost parts of the audio signal.
.Error Protection
adds error correcting codes to recover corrupted data.
4.Error Resilience
makes the source code more robust to transmission errors.
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