Hi, it will be interesting for you: git clone git://myau.su/recoded.git
25.12.2009, в 2:22, Álan Crístoffer написал(а): > 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>_______________________________________________ > libav-user mailing list > [email protected] > https://lists.mplayerhq.hu/mailman/listinfo/libav-user _______________________________________________ libav-user mailing list [email protected] https://lists.mplayerhq.hu/mailman/listinfo/libav-user
