Low Bit-Rate Coding: Codec Design

In contrast to the earlier codec techniques, the new codec designs are better suited to handle the digital audio systems. The most popular coding technique was linear pulse-code modulation (PMC), however it is not appropriate for all audio applications. The new codec was developed in response to the need for a reduced bit rate coding system. The industry has developed and grown as a result of the low bit rate coding processes, which are suited for various digital audio systems. Other codecs are sophisticated and only useful in the hands of professionals, while some are straightforward and simple to use. The new codecs have expanded the ways in which digital audio is used by transforming the storage and transmission applications.

Early codecs include the MUSICAM (masking pattern adopted universal subband integrated coding and multiplexing), OCF, and PXFM. The MUSICAM codec functions by deducting data based on subband analysis or psychoacoustic principles. The OCF and PXFM are both perceptual transform codecs and they have been advanced to include several aspects that allow them to serve the growing digital audio systems.

The moving picture expert group (MPEG) is one of the organizations whose contribution in digital audio transformation cannot be disputed. The group has devised several techniques of data reduction that allow the creation of audio and video materials. MPEG-1 standard operates in three parts including system, audio, and video. It supports audio sand video coding for CD playback within the CD’s bandwidth of 1.41 mbps. It also supports bit rates, stereo coding, and monaural coding. The MPEG bitstream format transmits data in frames and these frames can be decoded individually. Bit rates determine the length of each field and the content produced varies among layers.

MPEG-1 layer 1 is a simplified version of MUSICAM codec that uses subband coding. It provides high fidelity at a low complex. Further, it has a high bit rate. MPEG-1 LAYER II is a replica of the MUSICAM codec. Therefore, it is similar to layer I but with a highly complicated design. It has high fidelity, not very high complexity and moderate bit rates. MPEG-1 layer III is a refinement of ASPEC codec with some elements of MUSICAM. IT IS A TRANSFORM CODEC and it is more complex than layer 1 and II. It produces moderate fidelity even with low data rates and it is popularly known as MP3. MP3 uses a noise allocation system where bits are inherent outcomes of the strategy as opposed to allocating bits from SNR values. Hoffman and run-length entropy take advantage of numerical properties of the audio signal to curtail loss of data and generate large values at low frequencies, and smaller values at high frequencies. A variable bit rates has a higher transmission success, but the system transmitting the bit rate must also be variable.

MPEG-2 audio standard is used in internet downloading and HDTV transmission. It uses the advanced audio coding (AAC) format as well as the decoding mechanisms for MPEG-I. It, however, allows the use of incompatible codecs. MPEG-2 AAC produces high fidelity at low bit rates, but unlike the audio standard MPEG-2 ACC is not compatible with MPEG-1 because it is a non backward compatible coding. ATRAC codec is used to provide data reduction for the SDDS cinematic sound system. It has a 5.1 compression ratio, which is ideal for cinematic sound transmission. The perpetual audio coding codec is used to provide audio coding for digital televise and internet download purposes. It uses various mechanisms to achieve a reduction of 11.1 at the highest level of transparency. AC-3 codec and digital theatre system codec are also part of low bit-rate coding system.