Based On Audio-specific Dsp Core Of The Mpeg-2 Aac Decoder Design And Validation

AAC is a low rate, high-fidelity audio coding standard, which is characterized by high compression rate, good sound quality, supporting for multiple sampling rates and multiple channels. The maximal compression rate can be twenty to one. The maximum of channels are 48. Compared to MP3, AC-3 and other audio coding standards, AAC can effectively save bandwidth at a lower rate on the condition of the same sound quality. AAC is increasingly widely used in digital audio storage, multimedia network transmission, digital audio broadcasting and other fields. Therefore, the work of researching and implementing AAC decoder is of great significance.Now, the implementation of AAC decoder has two main methods, which are general-purpose DSP-based and ASIC implementation. In this paper, the AAC decoder is implemented based on audio-specific DSP core. This method is a tradeoff between above two methods. Compared to general-purpose DSP, the audio-specific DSP core has some special structures and special DSP instructions which are more suited to audio codecs. It can make the audio decoder meet the real-time requirement at a lower frequency. Compared to ASIC, the audio-specific DSP core is more flexible, it can be easily compatible with multiple audio standards and the design cycle based on it is shorter.This paper firstly analyzes the principle of each module in the MPEG-2 AAC decoder. To achieve the requirement of real-time decoding, Huffman decoding algorithm and inverse quantization algorithm are respectively optimized. Based on the characteristics of Huffman code table, a new fast Huffman decoding algorithm is proposed. The algorithm can fastly determine the code length of Huffman code word by using the first zero position of Huffman code word, and then acquires the position of the code word in the code table by using the code length and the initial code word of the code length. Then, the Huffman decoding can be completed fastly. The new Huffman decoding algorithm is faster than the algorithm of fixed-point C program by 30% increase; Anti-quantization algorithm is implemented by using the look-up table method and the polynomial approximation method. After algorithm optimization, the decoding program of AAC decoder is designed based on the instruction set of the audio-specific DSP core. Some techniques are applied in the process of designing the assembly program, such as arranging the code sequence reasonably, using the general-purpose register reasonably, reducing the function call and so on. Specially, IMDCT in the decoder is implemented by using DSP-specific instructions to call IMDCT hardware accelerator. Finally, AAC decoder is verified by hardware and software co-verification technology, and its performance is evaluated in three aspects which are decoding time, storage resource consumption and decoding quality. A piece of data, which is the most complicated in an AAC audio file and the sample rate of the audio file is 44.1KHz, is simulated by Modelsim. The decoding time of the piece of data is 233680 cycles. The storage resource consumption of the decoder is 52.35KB. The SNR of the AAC decoder is almost similar to business AAC decoder's. Through the experimental verification of AAC decoding, the decoder can work successfully that the main frequence of the decoder in the experiment is 33MHz, and the sound quality of the AAC decoder is good and there is no noise.