摘要 |
A method for an improved QSS (bit allocator) algorithm is disclosed. The disclosed method is capable of greatly improving determination time; thereby, improving the efficiency of converting a signal from an audio format to an MP3 format. The starting point of the QSS determination for a present frame (N) is the QSS of a previous frame (N-1). This starting point provides for improved efficiency for determining actual QSS of frame N as QSS[N-1] will be closer to QSS[N] than an arbitrary starting point. Thus, fewer iterations are required to determine QSS[N] as compared to conventional encoders. The algorithm of the present is more efficient than conventional methods in that it makes use of the fact that audio signal statistics usually do not change abruptly during the period of one audio frame to another. |