| 摘要 |
An audio codec and a method of compressing audio data makes use of a filterbank which automatically adapts itself to changes in the sampling frequency/bit rate to mimic the characteristics of the human auditory system. The algorithm used compares the bandwidth of each sub-band at a given depth with the critical bandwidth. If the critical bandwidth is less than the bandwidth of the sub-band, then the sub-band is split into two at the next level, and the process is repeated until the bandwidth of every sub-band is less than the critical bandwidth at the corresponding frequency. The codec thus automatically adapts itself to changes in sampling frequency/bit rate, which is particularly advantageous when very low bandwidths are in use. |
