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• fresh ffmpeg on 10.04 Could Not Find Codec Parameters

  26 juillet 2012, par Dmitry Kharlamov

    ffmpeg version git-2012-07-24-93342de Copyright (c) 2000-2012 the FFmpeg developers
  
    built on Jul 24 2012 23:55:41 with gcc 4.4.3 (Ubuntu 4.4.3-4ubuntu5.1)
  
    configuration: --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-swscale --enable-vdpau --enable-version3 --enable-libtheora --enable-libvorbis --enable-libvpx --enable-x11grab
  
    libavutil      51. 65.100 / 51. 65.100
  
    libavcodec     54. 44.100 / 54. 44.100
  
    libavformat    54. 20.100 / 54. 20.100
  
    libavdevice    54.  2.100 / 54.  2.100
  
    libavfilter     3.  3.100 /  3.  3.100
  
    libswscale      2.  1.100 /  2.  1.100
  
    libswresample   0. 15.100 /  0. 15.100
  
    libpostproc    52.  0.100 / 52.  0.100
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x2a1b240] Could not find codec parameters for stream 0 (Video: h264 (avc1 / 0x31637661), 1280x720): unspecified pixel format
  
  Consider increasing the value for the 'analyzeduration' and 'probesize' options
  
  /path/to/video.mp4: could not find codec parameters
  
  
  
  
  
  ffmpeg command line:
  
  /usr/local/bin/ffmpeg -i /path/to/video.mp4

• libavcodec avcodec_open2 returns -22

  28 juillet 2012, par buchtak

  I am trying to learn how to encode video using libavcodec library. I use the following initialization :

  avcodec_register_all();
  
  
  
  // This works fine.
  
  AVCodec *avcodec = avcodec_find_encoder( CODEC_ID_H264 );
  
  
  
  AVCodecContext *avctx = avcodec_alloc_context3( avcodec );
  
  avctx->bit_rate      = 400000;
  
  avctx->width         = 640;
  
  avctx->height        = 480;
  
  avctx->time_base.den = 15;
  
  avctx->time_base.num = 1;
  
  avctx->gop_size      = 10;
  
  avctx->max_b_frames  = 1;
  
  avctx->pix_fmt       = PIX_FMT_YUV420P;
  
  av_opt_set( avctx->priv_data, "preset", "slow", 0 );
  
  
  
  // ret should be zero, but it's negative
  
  int ret = avcodec_open2( avctx, avcodec, NULL );

  However, avcodec_open2(...) always returns a negative value. The avcodec_find_encoder(...) works fine and the returned pointer is not NULL. I use Win7 x64, 64-bit Zeranoe FFmpeg build from

  http://ffmpeg.zeranoe.com/builds/

  According to the readme the FFmpeg version is 2012-06-22 git-c17808c built with --enable-libx264. I also tried CODEC_ID_MPEG1VIDEO and changing some of the initialization parameters, but no matter what I do, the avcodec_open2(...) always returns value -22. The decoding/encoding example provided with the Zeranoe build (it's the same one as http://ffmpeg.org/doxygen/trunk/api-example_8c-source.html) does not work either...

• FFmpeg inaccurate outputs [closed]

  27 juillet 2012, par user1557780

  Possible Duplicate :
  ffmpeg : videos before and after conversion aren't the same length

  Recently, I've been trying to use FFmpeg for an application which requires a VERY accurate manipulation when it comes to the time parameter (milliseconds resolution). Unfortunately, I was surprised to find out that FFmpeg's manipulation functionalities return some inaccurate results.

  Here is the output of 'ffmpeg' :

  ffmpeg version 0.11.1 Copyright (c) 2000-2012 the FFmpeg developers
  
    built on Jul 25 2012 19:55:05 with gcc 4.2.1 (Apple Inc. build 5664)
  
    configuration: --enable-gpl --enable-shared --enable-pthreads --enable-libx264 --enable-libmp3lame
  
    libavutil      51. 54.100 / 51. 54.100
  
    libavcodec     54. 23.100 / 54. 23.100
  
    libavformat    54.  6.100 / 54.  6.100
  
    libavdevice    54.  0.100 / 54.  0.100
  
    libavfilter     2. 77.100 /  2. 77.100
  
    libswscale      2.  1.100 /  2.  1.100
  
    libswresample   0. 15.100 /  0. 15.100
  
    libpostproc    52.  0.100 / 52.  0.100

  Now, let's assume I want to rip the audio track of 'foo.mov'. Here is the relevant output of 'ffmpeg -i foo.mov' :

  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'foo.mov':
  
    Metadata:
  
      major_brand     : qt  
  
      minor_version   : 0
  
      compatible_brands: qt  
  
      creation_time   : 2012-07-24 23:16:08
  
    Duration: 00:00:40.38, start: 0.000000, bitrate: 805 kb/s
  
      Stream #0:0(und): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p, 480x360, 733 kb/s, 24.46 fps, 29.97 tbr, 600 tbn, 1200 tbc
  
      Metadata:
  
        rotate          : 90
  
        creation_time   : 2012-07-24 23:16:08
  
        handler_name    : Core Media Data Handler
  
      Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, mono, s16, 63 kb/s
  
      Metadata:
  
        creation_time   : 2012-07-24 23:16:08
  
        handler_name    : Core Media Data Handler

  As you probably noticed, the video file duration is 00:00:40.38. Using the following command, I ripped it's audio track :

  'ffmpeg -i foo.mov foo.wav'

  Output :

  Output #0, wav, to 'foo.wav':
  
    Metadata:
  
      major_brand     : qt  
  
      minor_version   : 0
  
      compatible_brands: qt  
  
      creation_time   : 2012-07-24 23:16:08
  
      encoder         : Lavf54.6.100
  
      Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s
  
      Metadata:
  
        creation_time   : 2012-07-24 23:16:08
  
        handler_name    : Core Media Data Handler
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (aac -> pcm_s16le)
  
  Press [q] to stop, [?] for help
  
  size=3482kB time=00:00:40.42 bitrate= 705.6kbits/s    
  
  video:0kB audio:3482kB global headers:0kB muxing overhead 0.001290%

  As you can see, the output file is longer than the file in the input.

  Another example is audio (and video) file trimming :
  Let's assume I would like to use ffmpeg for audio file trimming. I used the next command :

  'ffmpeg -t 00:00:10.000 -i foo.wav trimmed_foo.wav -ss 00:00:25.000'

  Output :

  [wav @ 0x10180e800] max_analyze_duration 5000000 reached at 5015510
  
  Guessed Channel Layout for  Input Stream #0.0 : mono
  
  Input #0, wav, from 'foo.wav':
  
    Duration: 00:00:40.42, bitrate: 705 kb/s
  
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s
  
  Output #0, wav, to 'trimmed_foo.wav':
  
    Metadata:
  
      encoder         : Lavf54.6.100
  
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (pcm_s16le -> pcm_s16le)
  
      Press [q] to stop, [?] for help
  
  size=864kB time=00:00:10.03 bitrate= 705.6kbits/s    
  
  video:0kB audio:864kB global headers:0kB muxing overhead 0.005199%

  Again, the output file is 30 milliseconds longer than I expected.

  I tried, for a long time, to research the issue without any success. When I use audacity for the same functionality, it does it very accurately !

  Does anyone have any idea how to solve this problem ?