Future is MPEG-4, Twelve fold compression achieved
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Recently a number of new technologies related to the storage capacity and the transmission bandwidth are emerging. From transmission part, for example, ADSL (Asymmetric Digital Subscriber Line) provides several Mbps transmission bandwidth for normal telephone line on down stream side. On storage part, hard disk capacity has being increased dramatically. A new high-density disc such as DVD (Digital Versatile Disc) also provides huge storage capability for audio and video sources. However, despite such tremendous growth of the storage capacity and the transmission bandwidth, the demand for higher quality of multimedia associated with audio, image, and video continues to outpace it. For instance, the required data rate satisfying the high-quality audio (more word size, more sample rate, and more channel) will be continuously increased, unless we give up to enjoy the digital audio world. Hence the importance of data compression is not likely to diminish, as a key technology to allow efficient storage and transmission.
The general idea behind data compression is to remove the redundancy present in the data to find more compact representations. Two families of algorithms exist in compression. When the information can be exactly recovered from the bits, the source coding or compression is called lossless; otherwise, it is called Lossy. To achieve higher compression ratios, lossy algorithms remove information from the original in a way that comes close to the original or that is not perceptible. In this case, therefore, we allow approximate representations of the original, instead of trying to represent the original exactly, and have only a modified version of the original after transmission. In contrast, lossless algorithms respect the integrity of the original signal. After transmission and reconstruction, an exact copy of the original signal is available.
The digital representation of audio data offers many advantages: high noise immunity, stability, and reproducibility. Audio in digital form also allows the efficient implementation of many audio processing functions (e.g., mixing, filtering, and equalization) through the digital computer. The conversion from the analog to the digital domain begins by sampling the audio input in regular, discrete intervals of time and quantizing the sampled values into a discrete number of evenly spaced levels. The digital audio data consists of a sequence of binary values representing the number of quantizer levels for each audio sample. The method of representing each sample with an independent codeword is called pulse code modulation (PCM).Compared to most digital data types the data rates associated with uncompressed digital audio are substantial. For example, the audio data on a compact disc (2 channels of audio sampled at 44.1 kHz with 16 bits per sample) requires a data rate of about 1.4 megabits per second. There is a clear need for some form of compression to enable the more efficient storage and transmission of this data.
The many forms of audio compression techniques differ in the trade-offs between encoder and decoder complexity, the compressed audio quality, and the amount of data compression. The techniques applied to general audio signals and are not specifically tuned for speech signals. The Motion Picture Experts Group (MPEG) audio compression algorithm is an International Organization for Standardization (ISO) standard for highfidelity audio compression. It is one part of a threepart compression standard. With the other two parts, video and systems, the composite standard addresses the compression of synchronized video and audio at a total bit rate of roughly 1.5 megabits per second.
The previous MPEG audio standards concentrate on the coding of audio signals with almost transparent quality. The MPEG-4 audio standard has been created to support different applications which range from intelligible speech to high quality multichannel audio. The MPEG-4 audio standardizes processing modules (tools) for natural and synthetic audio coding at bit rates ranging from 2 kbit/s up to 64 kbit/s. A single coding technique cannot accommodate both speech and audio at all desired bit rates. MPEG-4 also defines tools to synthesize sounds based on structured descriptions of audio data (also known as structured audio coding). Moreover, MPEG-4 provides bit rate scalability, complexity scalability and multi-bit rate operation.
The MPEG-4 systems layer facilitates the use of different tools and signaling this, and thus codecs according to existing standards can be accommodated. Each of the MPEG-4 coders is designed to operate in a stand-alone mode with its own bit stream syntax. Additional functionalities are realized both within individual coders, and by means of additional tools around the coders.
In our next discussion, we are going to explain every bit of compression technique that MPEG-4 uses. We designed this compression codec as a part of engineering degree and astonishingly we achieved compression up to twelve fold. We, at the end of this article series will also provide the code we have developed. So watch out for more.
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