An Efficient Method to Improve the Audio Quality Using AAC Low Complexity Decoder

This paper presents a new approach to design a Digital Audio Broadcast (DAB) audio decoder is introduced to improve the superiority of audio. Countries all over the world use DAB broadcasting systems more prominently, in Europe. DAB+ is the upgraded version of digital audio broadcasting. DAB and DAB+ coexist in many countries, so receivers are essential to be compatible with both standards. DAB+ is approximately twice as efficient as DAB due to the adoption of the AAC+ audio codec, and DAB+ can provide high quality audio with bit rates as low as 64 kbit/s. Integrating an MPEG-1 Layer II (MP2) decoder and Advanced Audio Coding Low Complexity (AAC LC) decoder provides a fundamental audio decoding for DAB and DAB+. The generated audio frames data from the DAB channel decoders are stored in RAM. The bit stream demultiplexer parses the quantized spectrum data in the audio. The inverse quantization performs the inverse quantization computation and synthesis filter generates the time domain Pulse Code Modulation (PCM) samples, all the above operation results writes them back to the audio RAM. The existing system of this project uses HE AAC V2 decoder, that system consists has SBR and PS technologies. This two technologies are used to improve the sound quality in low bit rate program. The proposed scheme is uses AAC LC and MP2 decoder it improve the sound quality in high bit rate. The simulation of this project is carried out by using MATLAB R2011a and Xilinx ISE 9.2i. DOI: 10.17762/ijritcc2321-8169.15039