Current existing digital audio signals are always restricted by sampling rates and bandwidth fit for the various storages and communication bandwidth. Take for example the widely spread mp3 tracks encoded by the standard MPEG1 layer 3. The audio bandwidth in MP3 is restricted to 16 kHz due to the protocols constraints defined. This paper presents the method to reconstruct the lost high frequency components from the bandlimited signals. Both the subjective and objective measures have been conducted and shown the better quality. Especially, the important objective measurement by the perceptual evaluation of audio quality system, which is the recommendation system by ITU-R Task Group 10/4 has proven a significant quality improvement.

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Main/Side (M/S) coding is a module deigned to reduce the channel irrelevancy and redundancies for multichannel coding in current audio standards. However, the success of this module relies on the careful design on four aspects: M/S decision, psychoacoustic model for the M/S channels, bits allocated between the M/S channels, and bit allocation in the M/S channels. This paper presents an efficient M/S method well designed from the four aspects. We conduct the experiments through the MP3 encoder to show the better quality and complexity than the Lame.

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The FIR-based reverberators, which convolve the input sequence with an impulse response modelling the concert hall, have better quality compared to the IIR-based approach. However, the high computational complexity of the FIR-based reverberators limits the applicability to most cost-oriented system. This paper introduces a method that uses perceptual criterion to reduce the complexity of convolution methods for reverberation. Also, an objective measurement criterion is introduced to check the perceptual difference from the reduction. The result has shown that the length of impulse response can be cut off by 60% without affecting the perceptual reverberation quality. The method is well integrated into the existing FFT-based approach is have around 30% speed-up.

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The bandwidth-proportional noise-shaping criterion has been proposed to be the optimum bit allocation criterion in the sense of minimizing segmental masking-to-noise ratio [1]. A single loop bit allocation has been developed for ISO/IEC MPEG1layer 3 (MP3) [2] to reduce the complexity over the existing two nested loop method for MP3 encoder. Also, the allocation method also leads to the better compression quality. This paper extends the allocation method to the ISO/IEC MPEG-2/4 Advanced Audio Coding (AAC) [3] [4]. This paper derives the perceptual bandwidth for the scale factor bands, provides the bit consumption trade-off to transmit the bits for the scale factors, and illustrates an efficient sectioning method on the Huffman codebook. All these methods jointly integrate with the bit allocation method to improve the quality and the computing complexity of the widely used AAC reference encoder, FAAC [5].

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