> On 14. Sep 2017, at 13:22, Anton Shekhovtsov <[email protected]> wrote: > > > > 2017-09-14 14:06 GMT+03:00 Info || Non-Lethal Applications > <[email protected] <mailto:[email protected]>>: > Hi guys, > > I just received a movie file with AAC audio which doesn't play correctly in > my player using the ffmpeg libraries. > It sounded “robotic” and the audio ran out of sync rather quickly. > > Ffprobe output: > > ffprobe version 3.3.1 Copyright (c) 2007-2017 the FFmpeg developers > built with llvm-gcc 4.2.1 (LLVM build 2336.11.00) > configuration: --prefix=/Volumes/Ramdisk/sw --enable-gpl --enable-pthreads > --enable-version3 --enable-libspeex --enable-libvpx --disable-decoder=libvpx > --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 > --enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb > --enable-filters --enable-libgsm --enable-libvidstab --enable-libx265 > --disable-doc --arch=x86_64 --enable-runtime-cpudetect > libavutil 55. 58.100 / 55. 58.100 > libavcodec 57. 89.100 / 57. 89.100 > libavformat 57. 71.100 / 57. 71.100 > libavdevice 57. 6.100 / 57. 6.100 > libavfilter 6. 82.100 / 6. 82.100 > libswscale 4. 6.100 / 4. 6.100 > libswresample 2. 7.100 / 2. 7.100 > libpostproc 54. 5.100 / 54. 5.100 > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from > '/Users/flo/Downloads/wetransfer-6ec47f/IQT Capture 170911 Unity Round1.mov': > Metadata: > major_brand : qt > minor_version : 512 > compatible_brands: qt > encoder : Lavf57.66.102 > Duration: 00:07:58.81, start: 0.000000, bitrate: 2672 kb/s > Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, > 1280x720, 2499 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default) > Metadata: > handler_name : DataHandler > Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, > fltp, 162 kb/s (default) > Metadata: > handler_name : DataHandler > > By examining the file I found out that the packet durations of the AAC stream > were not 1024 as they are with all the other files I have but rather either > 1000 or 1100. > While this by itself would not be a problem, it’s very confusing that the AAC > decoder outputs AVFrames with a duration of 1024. > > So now I’m a bit stuck as I really don’t know what to do. > If the packets were all 1000 I would try discarding the last 24 samples of > each AVFrame but as the frame apparently only holds 1024 samples although the > packet says it contains 1100 samples, I’m stuck. > I tried sending a NULL packet in the case of longer packets hoping to get the > remains 76 samples out of the decoder but I got this error message: > > [aac @ 0x10539da00] Got unexpected packet size after a partial decode > > The file plays fine in both QuickTime and VLC so there must be a correct way > of handling these kind of files.. > > I’d be very grateful for any insight anyone of you might have! > > Thanks and best, > > Flo > > _______________________________________________ > Libav-user mailing list > [email protected] <mailto:[email protected]> > http://ffmpeg.org/mailman/listinfo/libav-user > <http://ffmpeg.org/mailman/listinfo/libav-user> > > > What do you mean "packet contains 1100 samples", how is that specified? > AVPacket->duration is in time_base units, not in samples.
The audio stream has a time_base of 1/48000. So does the audio codec context. That’s what I find so weird.
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