Hi everyone! I am trying to get this puzzle solved, but no success so far. Can anyone please guide me further?
Thanks in advance for your time! On Fri, Jun 21, 2013 at 3:28 PM, Taha Ansari <mtaha.ans...@gmail.com> wrote: > > > > On Fri, Jun 21, 2013 at 3:06 PM, Carl Eugen Hoyos <ceho...@ag.or.at>wrote: > >> Taha Ansari <mtaha.ansari@...> writes: >> >> > I have run this application with existing mp4 >> > files as input, and it properly extracts audio, >> > and encodes to mp4 (audio only:AAC), or even >> > directly in AAC format (i.e. test.aac also >> > works). But when I tried running it on mp3 >> > files, output clip plays faster than it should >> > be (a clip of 1:12 seconds plays back till >> > 1:05 seconds only, and is also noisy). >> >> I did not look at your code but did you consider >> that the AAC decoder outputs AV_SAMPLE_FMT_FLTP >> and the MP3 decoder signed 16 bit values (I >> believe you can request planar or not)? >> >> Carl Eugen >> >> _______________________________________________ >> Libav-user mailing list >> Libav-user@ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/libav-user >> > > Hi Carl! > > As a matter of fact, I never knew about this, till now. In fact, when I > was probing the two files, I got s16 indication, so I thought they were > similar, maybe: > > > ---------------------------------------------------------------------------------------------------- > FFprobe from test.mp3 (input file): > > ---------------------------------------------------------------------------------------------------- > ffprobe version N-47062-g26c531c Copyright (c) 2007-2012 the FFmpeg > developers > built on Nov 25 2012 12:23:20 with gcc 4.7.2 (GCC) > configuration: --disable-static --enable-shared --enable-gpl > --enable-version3 > --disable-pthreads --enable-runtime-cpudetect --enable-avisynth > --enable-bzlib > --enable-frei0r --enable-libass --enable-libopencore-amrnb > --enable-libopencore- > amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame > --enable-libnut - > -enable-libopenjpeg --enable-libopus --enable-librtmp > --enable-libschroedinger - > -enable-libspeex --enable-libtheora --enable-libutvideo > --enable-libvo-aacenc -- > enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 > --enab > le-libxavs --enable-libxvid --enable-zlib > libavutil 52. 9.100 / 52. 9.100 > libavcodec 54. 77.100 / 54. 77.100 > libavformat 54. 37.100 / 54. 37.100 > libavdevice 54. 3.100 / 54. 3.100 > libavfilter 3. 23.102 / 3. 23.102 > libswscale 2. 1.102 / 2. 1.102 > libswresample 0. 17.101 / 0. 17.101 > libpostproc 52. 2.100 / 52. 2.100 > [mp3 @ 007b2a60] max_analyze_duration 5000000 reached at 5015510 > Input #0, mp3, from 'test.mp3': > Metadata: > major_brand : isom > minor_version : 512 > compatible_brands: isomiso2avc1mp41 > encoder : Lavf54.37.100 > Duration: 00:01:12.67, start: 0.000000, bitrate: 128 kb/s > Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16, 128 kb/s > > ---------------------------------------------------------------------------------------------------- > > > ---------------------------------------------------------------------------------------------------- > FFprobe from test.mp4 (converted file): > > ---------------------------------------------------------------------------------------------------- > > ffprobe version N-47062-g26c531c Copyright (c) 2007-2012 the FFmpeg > developers > built on Nov 25 2012 12:23:20 with gcc 4.7.2 (GCC) > configuration: --disable-static --enable-shared --enable-gpl > --enable-version3 > --disable-pthreads --enable-runtime-cpudetect --enable-avisynth > --enable-bzlib > --enable-frei0r --enable-libass --enable-libopencore-amrnb > --enable-libopencore- > amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame > --enable-libnut - > -enable-libopenjpeg --enable-libopus --enable-librtmp > --enable-libschroedinger - > -enable-libspeex --enable-libtheora --enable-libutvideo > --enable-libvo-aacenc -- > enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 > --enab > le-libxavs --enable-libxvid --enable-zlib > libavutil 52. 9.100 / 52. 9.100 > libavcodec 54. 77.100 / 54. 77.100 > libavformat 54. 37.100 / 54. 37.100 > libavdevice 54. 3.100 / 54. 3.100 > libavfilter 3. 23.102 / 3. 23.102 > libswscale 2. 1.102 / 2. 1.102 > libswresample 0. 17.101 / 0. 17.101 > libpostproc 52. 2.100 / 52. 2.100 > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test.mp4': > Metadata: > major_brand : isom > minor_version : 512 > compatible_brands: isomiso2mp41 > encoder : Lavf54.37.100 > Duration: 00:01:04.62, start: 0.000000, bitrate: 129 kb/s > Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, > s16, 128 > kb/s > Metadata: > handler_name : SoundHandler > > ---------------------------------------------------------------------------------------------------- > > Hence the reason I was supplying: > > c->sample_fmt = AV_SAMPLE_FMT_S16; (in add_audio_stream() function). > > If I'm not wasting too much of your time, can you please guide how I can > co relate the two formats, pragmatically? > > Thanks for your time! >
_______________________________________________ Libav-user mailing list Libav-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/libav-user