Another problem with this codec is that it is an enormous hack. In particular, it sets the nBlockAlign member of the wave format to 1. This means that applications think a single byte can be decompressed into audio, when in fact the Fraunhofer-IIS codec will buffer up data until it has enough to decompress a Layer 3 frame. Extracted sections of such audio streams will often have muted tails because the audio codec discards fractions of frames at the start. The root of these problems is the MP3 spec, which allows audio frames to "borrow" unused bandwidth from earlier frames, making it impossible for the MP3 audio codec to specify a fixed size for a self-contained audio block.
Finally, the Fraunhofer-IIS codec is very lazy and incorrectly sets the bitrate of the stream. For instance, a 48Kbit stream encoded by the Fraunhofer-IIS codec has a specified rate of 6000 bytes/sec, when in reality the stream is about 5971 bytes/sec. This 0.0048% difference may not seem like much, except that it causes the audio to race past the video approximately one second for every 200 seconds of video. One way to ‘fix’ this problem is to correspondingly adjust the video frame rate to compensate. VirtualDub will automatically correct for this problem when compressing audio to MPEG format.
J'ai pas dit que le codec original était nul, j'ai dit que le codec rippé avait un problème