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!
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