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• FFmpeg ogg - Invalid data found when processing input

  28 juillet 2017, par Earinor

  I am trying to change the framerate of a Video of mine. Below there is the example Input and the Output. I Built ffmpeg myself and I’m pretty sure I miss the right decoder/demuxer but I have no Idea what I need. I thought —enable-libtheora would be enough to enable, muxers, demuxers, encoder and decoder, but apperently there is somehting missing. Does anybody have an idea about this ?

  I have inserted path-to-ffmpeg to make this shorter, the Path itself is okay, so no worries about that.

     *path-to-ffmpeg*\ffmpeg>ffmpeg -i Video.ogg -filter "minterpolate='fps=60'" out.ogg
  
  ffmpeg version 3.3.git Copyright (c) 2000-2017 the FFmpeg developers
  
    built with gcc 6.3.0 (x86_64-posix-seh-rev2, Built by MinGW-W64 project)
  
    configuration: --disable-encoders --disable-decoder=hevc --disable-decoder=raw
  
  video --disable-decoder=ac3 --disable-decoder=flac --disable-decoder=ffwavesynth
  
   --disable-decoder=libcelt --disable-decoder=libgsm --disable-decoder=libilbc --
  
  disable-decoder=libopus --disable-decoder=dvbsub --disable-decoder=dvdsub --disa
  
  ble-hwaccels --disable-parsers --disable-bsfs --disable-protocols --disable-inde
  
  vs --disable-outdevs --disable-devices --disable-filters --disable-muxers --disa
  
  ble-demuxers --enable-muxer=ogg --enable-zlib --enable-libtheora --enable-libvor
  
  bis --enable-shared --enable-encoder=libtheora --enable-encoder=libvorbis --enab
  
  le-demuxer=image2 --enable-filter=framerate --enable-protocol=file --enable-filt
  
  er=minterpolate --enable-filter=scale
  
    libavutil      55. 61.100 / 55. 61.100
  
    libavcodec     57. 93.100 / 57. 93.100
  
    libavformat    57. 72.101 / 57. 72.101
  
    libavdevice    57.  7.100 / 57.  7.100
  
    libavfilter     6. 85.100 /  6. 85.100
  
    libswscale      4.  7.101 /  4.  7.101
  
    libswresample   2.  8.100 /  2.  8.100
  
  Video.ogg: Invalid data found when processing input

• How to link libav* libraries of FFmpeg in my own program ?

  20 août 2019, par zuguorui

  I am new to FFmpeg, now I want to use the FFmpeg codec compenet in my own program. I’ve followed the guide on FFmpeg wiki to complie the libraries the FFmpeg and get them done. But when I compile my own program it always tips ld: library not found for -lavutil or something like this.

  Environment：MacOS 10.14.6

  Compile tool : clang++

  steps :

  (1)Install all needed libraries follow the wiki.

  (2)./configuure --prefix=host --enable-shared --disable-static

  (3)make

  (4)make install, after this step, I got compiled files in dictionary host.

  (5)copy the source code in FFmpeg/doc/examples/decode_audio.c to create a new project, and copy includes and dylib files to my project dictionary. The structure of my project is :

  ./
  
  |----FFmpeg
  
       |----include
  
            |----libavcodec
  
                 |----*headers*
  
            |----libavformat
  
                 |----*headers*
  
            |----libavutil
  
                 |----*headers*
  
       |----lib
  
            |----libavutil.dylib
  
            |----libavcodec.dylib
  
            |----libavformat.dylib
  
  |----main.cpp

  (6)compile the project.

  clang++ *.cpp -L./FFmpeg/lib/ -lavutil -lavformat -lavcodec -I ./FFmpeg/include -o main.out

  But it always fail with tip ld: library not found for -lavutil or something like this. I don’t know why. Do I need to install those libraries in /usr/local dictionary instead of host ? Or I miss some variables when configing or compiling ?

  update：I solved this problem, I copyed the links of dylib files but not dylib itself, which caused this problem. But a new problem rised, when I run the program, it tips dyld: Library not loaded: @executable_path/lib/libavcodec.dylib. Reason: image not loaded.

• Unable to find audio stream

  18 novembre 2011, par Kurt

  I'm trying to adapt a code in order to add a sound capture feature (on a live stream), with the help of ffmpeg and directshow.

  When i try to play with ffplay the AVIsynth file, everything works perfectly i've got the audio and video. But when i open this input file by code i only find the video stream.

  The Avs file :

  V = DirectShowSource("Decklink_HDMI.grf", fps=10, framecount=1000000000, seek=false, audio=false)
  
  A = DirectShowSource("Decklink_Audio.grf", fps=1, framecount=1000000000, video=false)
  
  AudioDub(V, A)

  The opening code :

  ffmpeg::AVInputFormat * ifmt;
  
  
  
  ifmt = ffmpeg::av_find_input_format("avs");
  
  
  
  // Open input file
  
   if(ffmpeg::avformat_open_input(&pFormatCtx, filename.toStdString().c_str(), ifmt, NULL) != 0)

  When i make a variable lookout on gdb just after the opening.

  i'm looking at nb_streams in pFormatCtx->nb_streams and it's at 1
  The only stream i can find in pFormatCtx->stream is a video one. And that's why i'm not able to capture the sound.

  So i'm asking if i'm doing something wrong with my opening or if i miss something.

  Ps : I know the code shown is short but the problem appears at this very step !

  Thank you

  Kurt

  — EDIT —

  I've also noticed that when I dump the AVFormatContext the video stream got a 456x32 size.
  And i can find this very same size of the window displayed when i try to launch a corrupted script with ffplay.

  The original video format when i play the correct script with ffplay is of 1920x1080

  I think my problem is maybe deeper than the simple fact of not being able to get the audio stream.

  I'm trying to find out how to know the error message that is displayed on this 456x32 windows

  — EDIT2 —

  I Find out what is written on this image and my problem is solved, badly placed avs script an old one was is place.

  I'm ashamed.