Yes, the problem was some wrong value in avcodec_encode_audio(), now sound is ok when converting one mp3 into another.
But I can't resample the audio, I don't know how to get the number of samples decoded... none of the ways I found in sample codes worked (the buffer is not filled in, but execution continues normally). On Tue, Dec 22, 2009 at 17:58, Álan Crístoffer <[email protected]>wrote: > I came to find out that the problem is not with pts/dts, but with my > packets having lots of silence between samples, > going to take a look in the sizes I'm passing to the encode and write > functions... > > > On Tue, Dec 22, 2009 at 16:58, Álan Crístoffer <[email protected]>wrote: > >> Hi again, >> >> I'm trying to write a class to simplify using ffmpeg, but I having some >> problems: >> >> VIDEO: >> >> 1. Runs twice faster than original file >> 2. Poor quality in image >> >> AUDIO: >> >> 1. when encoding as MP3, it has poor quality ( noises and wrong volume >> ) >> 2. when encoding as OGG, the file of 30 secs becomes a ~2min14sec >> 3. when encoding as WAV, the file of 30 secs becomes a 46min56secs! >> >> I'm not working in the video stuff right now (did rewrite my class 3 times >> already, starting from sound this time), >> so if you could explain me how to get the pts/dts done right (i'm not >> setting it at all, letting it for the packet to handle) >> I would be very glad :) >> I did already take a look at various sample codes, but wich one does >> things in a different way... >> >> >> thanks, >> -- >> Álan Crístoffer >> > > > > -- > Álan Crístoffer > -- Álan Crístoffer _______________________________________________ libav-user mailing list [email protected] https://lists.mplayerhq.hu/mailman/listinfo/libav-user
