In this paper, we describe a simple method for reproducing high frequency components at low-bit rate audio coding. To compress an audio signal at low-bit rates (below 16 kbps per channel) we should use a lower sampling frequency (below 16 kHz) or high performance audio coding technology. When an audio signal is sampled at a low frequency and coded at a low-bit rate, high frequency components are lost and reverberant sound because of quantization noise between pitch pulses. In short-term period, the harmonic characteristic of audio signal is stationary, so the replication of high-frequency bands with low-frequency bands can extend the frequency range of resulting sound and enhance the sound quality. In addition, for reducing the number of bands to be reproduced we adapt this algorithm at the Bark scale domain. For the compatibility with conventional audio decoder, the additional bit stream is added at the end of each frame, which is generated by conventional audio coder. We adapt this proposed algorithm to MPEG-2 AAC and confirmed to increase the quality of audio in comparison with the conventional MPEG-2 AAC coded audio at the same rate. The computational cost of proposed algorithm is similar or a little more than conventional MPEG-2 AAC decoder.
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