hello, I'm having problems when converting audio files. When I convert a mp3 (or ogg, or flac, or mov) into a mp3(or ogg, or flac, or mov (keep the format)) of same sample rate and channels numbers (in other words, no resampling) everything is ok.
when I resample, the audio output has poor quality, and when I convert from one format to other the result is undefined (poor quality, big files, or a mix of the two). I did already see (using Audacity) that the problem is "blank spaces" in the stream, as if encoding was putting a size bigger than the real data size, so each packet is written in disk with some extra null data, that generates silence that shouldn't be there (the cause of poor quality). This behavior is clearly observed when encoding as WAV and looking at the file with Audacity ( a little zoom shows you the "blank spaces" in the file after each packet) the project is attached (Qt needed). thanks, -- Álan Crístoffer
LibAVConverter.tar.bz2
Description: BZip2 compressed data
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