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mp3DirectCut Question

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#1
Sometimes after I use mp3DirectCut and save my edit I get a synch error. One was even 50 errors. Yet when I listened it sounded fine. What exactly does synch error mean and will it impact my enjoyment of my mp3's>
Thanks.
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#2
Mp3 uses frames at a constant bitrate (CBR) or frames at a few different bit rates (VBR,ABR). To make it more flexible (=less bits for easy to encode parts, more for hard ones) there's a feature called bit reservoir.
If the encoder assumes that not all bits of a frame are needed at a certain position, these bits can be "saved" and used for storing information from the next frame.
If cut in a position where bit reservoir is big, some information is missing at the beginning of the part after the cut resulting in a "synch error".

If you have the original CD(s) you can avoid this by editing uncompressed .wav before encoding. In lame there's also a --nores switch which disables bit reservoir, but it will decrease overall quality.

I don't think there's a way to fix this if you don't have a choice what mp3s to use, but maybe you might want to ask this at http://www.hydrogenaudio.org forums. - Many knowledgable helpful people there ...
This message has been edited since its posting. Latest edit was made on 18 Sep 2003 @ 11:49

#3
What I'm doing is editing the first dead air part of mp3s. I guess my real concern is will these synch errors affect how my mp3s sound. Maybe cutting off the first part is producing the errors and may have no effect. What do you think?
#4
The decoded data will be affected at the very beginning, something like ~ 1 second at most. Sometimes the sound there will be changed audibly, sometimes not. If I didn't want to spend time on checking (= listening to) every song seperately, I would just cut ~ 1 second before the music starts.
Quote:
What do you think?
I think ...
- this is a good, interesting question.
- I don't know enough about mp3 format to give a really good answer.
- some people hydrogenaudio (link above) are programmers with deep knowledge in audio compression, there are even lame developers. I recommend to ask your question there.
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