| On the audio perceptual coding technology, research mainly concentrates in two aspects: first, optimization of audio compression algorithms; second, hardware design and implementation of the algorithms. At present, with the popularity of mobile network, the spread of audio products gets more frequent and convenient. Due to the limitation of computing capability and storage capacity which comes along with the mobile terminal, realization of an audio coding system with the performance of low complexity and high quality has become one of the most popular researches in digital audio processing.To achieve a high-quality audio codec with low complexity, this paper focused on two improvements: first, the key technologies of AAC, such as frequency transform, psychoacoustic analysis and quantization, on the algorithm level were optimized, in order to reduce computational complexity; second, based on the SOPC design strategy, a real-time MPEG AAC codec system was implemented using the combination of soft-core microprocessor and IP cores.The main work and innovation are as follows:(1) The filterbank module is a computation-intense part of audio perceptual coding model, occupying large amount of computing. In this paper, two methods accelerating the computation speed of the filterbank are proposed. One method was based on a recursive structure and the other was with the N/8-point FFT kernel, which were suitable for IP core design of both MDCT and IMDCT. Compared with the other recursive algorithms, the first approach reduced its computation cycles to N~2/16 and provided a superior performance in terms of computation speed, data throughput and hardware utilization. Although the existing algorithms based on N/4-point FFT kernel cost fewer adders, the second method not only cut down the requirement of multipliers, but also doubled the computation rate.(2) To eliminate the impact of pre-echo, the psychoacoustic module in audio perceptual coding model adopts transient analysis method to switch adaptively the transformation length. Based on the characteristics of human auditory and audio compression technique, a hypothesis of perceptual threshold model is presented. Besides, a block switching method in time domain is exploited instead of PE-based algorithm, which could quickly determine the transient signal. As a result, it raised the computing speed of the psychoacoustic model with little effect on the audio quality.(3) The quantization module of audio coding system employs Brandenburg architecture to obtain good quality, but it results in great complexity, which is not suitable for real-time applications. A simplification of the dual-loop structure is proposed on the basis of the noise approximate formula. According to the relation between the common scalefactor and the scalefactors in each scalefactor bands, the iterating scope of quantizing step got narrower to expedite its convergence. Results of experiments showed that the quality of reconstructed sound with the proposed approach was almost the same as the one reconstructed by original quantization module. In the decoding system, a modified version of a look-up table method is exhibited to perform the inverse non uniform quantization. In comparison with the existing ones, it reduced 50% storage and decreased the calculation errors.(4) A programmable model of digital audio codec is developed with the concept of SOPC architecture. Taking MPEG AAC as an example, the software/hardware co-operation was processed to reduce the computational complexity of the codec system. The reports of FPGA implementation showed that this audio codec system achieved higher coding rate and realized real-time decoding procedure. The results of both objective and subjective evaluation tests indicated this codec got good audio quality.
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