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• FFMPEG : concat demuxer errors during execution

  1er novembre 2015, par lgaravaglia

  I’m trying to use the FFMPEG concat demuxer to create a "loop" of a single video.

  I first tried running this command :

  ffmpeg -f concat -i behzad.txt -c:v copy output_concat.avi

  My input text file is as follows :

  file 'G:\Masters_Thesis_Files\Honda_Database\Database1\Training\videos\behzad\behzad.avi'
  
  file 'G:\Masters_Thesis_Files\Honda_Database\Database1\Training\videos\behzad\behzad.avi'

  The input file is in raw format.

  After running the command I get the following output to the console and the video generated is just the first frame of my input video file.

  ffmpeg version N-74286-ge5774f2 Copyright (c) 2000-2015 the FFmpeg developers
  
    built with gcc 4.9.3 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
  
  isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
  
  le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --
  
  enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l
  
  ibilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab
  
  le-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --en
  
  able-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --ena
  
  ble-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc
  
   --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enabl
  
  e-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-lzma --ena
  
  ble-decklink --enable-zlib
  
    libavutil      54. 30.100 / 54. 30.100
  
    libavcodec     56. 57.100 / 56. 57.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 32.100 /  5. 32.100
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, concat, from 'behzad.txt':
  
    Duration: N/A, start: 0.000001, bitrate: 110806 kb/s
  
      Stream #0:0: Video: rawvideo, bgr24, 640x480, 110806 kb/s, 14.99 fps, 14.99
  
  tbr, 14.99 tbn, 14.99 tbc
  
  File 'output_concat.avi' already exists. Overwrite ? [y/N] y
  
  Output #0, avi, to 'output_concat.avi':
  
    Metadata:
  
      ISFT            : Lavf56.40.101
  
      Stream #0:0: Video: rawvideo, bgr24, 640x480, q=2-31, 110806 kb/s, 14.99 fps
  
  , 14.99 tbr, 14.99 tbn, 14.99 tbc
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  [avi @ 0000000004723360] Non-monotonous DTS in output stream 0:0; previous: 1186
  
  24821151886176, current: 118624821151886176; changing to 118624821151886177. Thi
  
  s may result in incorrect timestamps in the output file.
  
  [avi @ 0000000004723360] Too large number of skipped frames 118624821151886176 >
  
   60000
  
  av_interleaved_write_frame(): Invalid argument
  
  [avi @ 0000000004723360] Too large number of skipped frames 118624821151886176 >
  
   60000
  
  frame=    2 fps=0.0 q=-1.0 Lsize=     906kB time=-00:00:00.00 bitrate=N/A
  
  video:1800kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing
  
  overhead: unknown
  
  Conversion failed!

  I also tried to run the command with the codec set to h264, rather than just copying the codec. When I ran this command the output to the console still had the DTS errors, however, my output video contained the entire input video rather than just the first frame. It did not repeat the video like I expect.

  I should also specify, I’m running these commands in Windows using the latest version of Zeranoe FFMPEG.

  I’m still relatively inexperienced with FFMPEG, is there some part of the process that I’m doing wrong ?

• ffmpeg amerge and amix filter delay

  10 février 2016, par User New

  I need to take audio-streams from several IP cameras and merge them into one file, so that they would sound simaltaneousely.

  I tried filter "amix" : (for testing purposes I take audio-stream 2 times from the same camera. yes, I tried 2 cameras - result is the same)

  ffmpeg -i rtsp://user:pass@172.22.5.202 -i rtsp://user:pass@172.22.5.202 -map 0:a -map 1:a  -filter_complex amix=inputs=2:duration=first:dropout_transition=3  -ar 22050 -vn -f flv rtmp://172.22.45.38:1935/live/stream1

  result : I say "hello". And hear in speakers the first "hello" and in 1 second I hear the second "hello". Instead of hearing two "hello"’s simaltaneousely.

  and tried filter "amerge" :

  ffmpeg -i rtsp://user:pass@172.22.5.202 -i rtsp://user:pass@172.22.5.202 -map 0:a -map 1:a  -filter_complex amerge -ar 22050 -vn -f flv rtmp://172.22.45.38:1935/live/stream1

  result : the same as in the first example, but now I hear the first "hello" in left speaker and in 1 second I hear the second "hello" in right speaker, instead of hearing two "hello"’s in both speakers simaltaneousely.

  So, the question is : how to make them sound simaltaneousely ? May be you know some parameter ? or some other command ?

  P.S. Here is ful command-line output for both variants if you need them :
  amix :

  [root@minjust ~]# ffmpeg -i rtsp://admin:12345@172.22.5.202 -i rtsp://admin:12345@172.22.5.202 -map 0:a -map 1:a -filter_complex amix=inputs=2:duration=longest:dropout_transition=0 -vn -ar 22050 -f flv rtmp://172.22.45.38:1935/live/stream1       ffmpeg version N-76031-g9099079 Copyright (c) 2000-2015 the FFmpeg developers
  
    built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-16)
  
    configuration: --enable-gpl --enable-libx264 --enable-libmp3lame --enable-nonfree --enable-version3
  
    libavutil      55.  4.100 / 55.  4.100
  
    libavcodec     57.  6.100 / 57.  6.100
  
    libavformat    57.  4.100 / 57.  4.100
  
    libavdevice    57.  0.100 / 57.  0.100
  
    libavfilter     6. 11.100 /  6. 11.100
  
    libswscale      4.  0.100 /  4.  0.100
  
    libswresample   2.  0.100 /  2.  0.100
  
    libpostproc    54.  0.100 / 54.  0.100
  
  Input #0, rtsp, from 'rtsp://admin:12345@172.22.5.202':
  
    Metadata:
  
      title           : Media Presentation
  
    Duration: N/A, start: 0.032000, bitrate: N/A
  
      Stream #0:0: Video: h264 (Baseline), yuv420p, 1280x720, 20 fps, 25 tbr, 90k tbn, 40 tbc
  
      Stream #0:1: Audio: adpcm_g726, 8000 Hz, mono, s16, 16 kb/s
  
      Stream #0:2: Data: none
  
  Input #1, rtsp, from 'rtsp://admin:12345@172.22.5.202':
  
    Metadata:
  
      title           : Media Presentation
  
    Duration: N/A, start: 0.032000, bitrate: N/A
  
      Stream #1:0: Video: h264 (Baseline), yuv420p, 1280x720, 20 fps, 25 tbr, 90k tbn, 40 tbc
  
      Stream #1:1: Audio: adpcm_g726, 8000 Hz, mono, s16, 16 kb/s
  
      Stream #1:2: Data: none
  
  Output #0, flv, to 'rtmp://172.22.45.38:1935/live/stream1':
  
    Metadata:
  
      title           : Media Presentation
  
      encoder         : Lavf57.4.100
  
      Stream #0:0: Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002), 22050 Hz, mono, fltp (default)
  
      Metadata:
  
        encoder         : Lavc57.6.100 libmp3lame
  
  Stream mapping:
  
    Stream #0:1 (g726) -> amix:input0
  
    Stream #1:1 (g726) -> amix:input1
  
    amix -> Stream #0:0 (libmp3lame)
  
  Press [q] to stop, [?] for help
  
  [rtsp @ 0x2689600] Thread message queue blocking; consider raising the thread_queue_size option (current value: 8)
  
  [rtsp @ 0x2727c60] Thread message queue blocking; consider raising the thread_queue_size option (current value: 8)
  
  [rtsp @ 0x2689600] max delay reached. need to consume packet
  
  [NULL @ 0x268c500] RTP: missed 38 packets
  
  [rtsp @ 0x2689600] max delay reached. need to consume packet
  
  [NULL @ 0x268d460] RTP: missed 4 packets
  
  [flv @ 0x2958360] Failed to update header with correct duration.
  
  [flv @ 0x2958360] Failed to update header with correct filesize.
  
  size=      28kB time=00:00:06.18 bitrate=  36.7kbits/s
  
  video:0kB audio:24kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 16.331224%

  and amerge :

  [root@minjust ~]# ffmpeg -i rtsp://admin:12345@172.22.5.202 -i rtsp://admin:12345@172.22.5.202 -map 0:a -map 1:a -filter_complex amerge -vn -ar 22050 -f flv rtmp://172.22.45.38:1935/live/stream1
  
  ffmpeg version N-76031-g9099079 Copyright (c) 2000-2015 the FFmpeg developers
  
    built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-16)
  
    configuration: --enable-gpl --enable-libx264 --enable-libmp3lame --enable-nonfree --enable-version3
  
    libavutil      55.  4.100 / 55.  4.100
  
    libavcodec     57.  6.100 / 57.  6.100
  
    libavformat    57.  4.100 / 57.  4.100
  
    libavdevice    57.  0.100 / 57.  0.100
  
    libavfilter     6. 11.100 /  6. 11.100
  
    libswscale      4.  0.100 /  4.  0.100
  
    libswresample   2.  0.100 /  2.  0.100
  
    libpostproc    54.  0.100 / 54.  0.100
  
  Input #0, rtsp, from 'rtsp://admin:12345@172.22.5.202':
  
    Metadata:
  
      title           : Media Presentation
  
    Duration: N/A, start: 0.064000, bitrate: N/A
  
      Stream #0:0: Video: h264 (Baseline), yuv420p, 1280x720, 20 fps, 25 tbr, 90k tbn, 40 tbc
  
      Stream #0:1: Audio: adpcm_g726, 8000 Hz, mono, s16, 16 kb/s
  
      Stream #0:2: Data: none
  
  Input #1, rtsp, from 'rtsp://admin:12345@172.22.5.202':
  
    Metadata:
  
      title           : Media Presentation
  
    Duration: N/A, start: 0.032000, bitrate: N/A
  
      Stream #1:0: Video: h264 (Baseline), yuv420p, 1280x720, 20 fps, 25 tbr, 90k tbn, 40 tbc
  
      Stream #1:1: Audio: adpcm_g726, 8000 Hz, mono, s16, 16 kb/s
  
      Stream #1:2: Data: none
  
  [Parsed_amerge_0 @ 0x3069cc0] No channel layout for input 1
  
  [Parsed_amerge_0 @ 0x3069cc0] Input channel layouts overlap: output layout will be determined by the number of distinct input channels
  
  Output #0, flv, to 'rtmp://172.22.45.38:1935/live/stream1':
  
    Metadata:
  
      title           : Media Presentation
  
      encoder         : Lavf57.4.100
  
      Stream #0:0: Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002), 22050 Hz, stereo, s16p (default)
  
      Metadata:
  
        encoder         : Lavc57.6.100 libmp3lame
  
  Stream mapping:
  
    Stream #0:1 (g726) -> amerge:in0
  
    Stream #1:1 (g726) -> amerge:in1
  
    amerge -> Stream #0:0 (libmp3lame)
  
  Press [q] to stop, [?] for help
  
  [rtsp @ 0x2f71640] Thread message queue blocking; consider raising the thread_queue_size option (current value: 8)
  
  [rtsp @ 0x300fb40] Thread message queue blocking; consider raising the thread_queue_size option (current value: 8)
  
  [rtsp @ 0x2f71640] max delay reached. need to consume packet
  
  [NULL @ 0x2f744a0] RTP: missed 18 packets
  
  [flv @ 0x3058b00] Failed to update header with correct duration.
  
  [flv @ 0x3058b00] Failed to update header with correct filesize.
  
  size=      39kB time=00:00:04.54 bitrate=  70.2kbits/s
  
  video:0kB audio:36kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 8.330614%

  Thanx.

  UPDATE 30 oct 2015 : I found interesting detail when connecting 2 cameras (they have different microphones and I hear the difference between them) : the order of "Hello"’s from different cams depends on the ORDER OF INPUTS.
  with command

  ffmpeg -i rtsp://cam2 -i rtsp://cam1 -map 0:a -map 1:a -filter_complex amix=inputs=2:duration=longest:dropout_transition=0 -vn -ar 22050 -f flv rtmp://172.22.45.38:1935/live/stream1

  I hear "hello" from 1st cam and then in 1 second "hello" from 2nd cam.

  with command

  ffmpeg -i rtsp://cam1 -i rtsp://cam2 -map 0:a -map 1:a -filter_complex amix=inputs=2:duration=longest:dropout_transition=0 -vn -ar 22050 -f flv rtmp://172.22.45.38:1935/live/stream1

  I hear "hello" from 2nd cam and then in 1 second "hello" from 1st cam.

  So, As I understand - ffmpeg takes inputs not simaltaneousely, but in the order of inputs given.
  Question : how to tell ffmpeg to read inputs simaltaneousely ?

• gdb not stopping at even trivial breakpoints

  18 novembre 2015, par baruch

  I am trying to debug ffplay (a debug build) compiled using mingw-w64 on msys2 on Windows 7 64bit. I tried setting breakpoints using gdb, and they seem to be set fine, but the program does not stop. So i tried issuing start, and again it didn’t stop. I tried manually setting a breakpoint at main, but it still didn’t stop. I verified that main is found, and it is. Here is a portion of the session. Any explanation what might cause this ?

  (gdb) start
  
  Temporary breakpoint 1 at 0x140013f7c: file C:/Users/bubaruch/Documents/projects/ffmpeg/ffmpeg/ffplay.c, line 3756.
  
  Starting program: C:\Users\bubaruch\Documents\projects\ffmpeg\b2\ffplay_g.exe
  
  [New Thread 7752.0x275c]
  
  ffplay version 2.8.git Copyright (c) 2003-2015 the FFmpeg developers
  
    built with gcc 5.2.0 (Rev4, Built by MSYS2 project)
  
    configuration: --disable-doc --disable-optimizations
  
    libavutil      55.  5.100 / 55.  5.100
  
    libavcodec     57. 15.100 / 57. 15.100
  
    libavformat    57. 14.100 / 57. 14.100
  
    libavdevice    57.  0.100 / 57.  0.100
  
    libavfilter     6. 15.100 /  6. 15.100
  
    libswscale      4.  0.100 /  4.  0.100
  
    libswresample   2.  0.101 /  2.  0.101
  
  Simple media player
  
  usage: ffplay [options] input_file
  
  
  
  An input file must be specified
  
  Use -h to get full help or, even better, run 'man ffplay'
  
  [Inferior 1 (process 7752) exited with code 01]
  
  
  
  (gdb) l main
  
  3748       return 1;
  
  3749    }
  
  3750
  
  3751    /* Called from the main */
  
  3752    int main(int argc, char **argv)
  
  3753    {
  
  3754        int flags;
  
  3755        VideoState *is;
  
  3756        char dummy_videodriver[] = "SDL_VIDEODRIVER=dummy";
  
  3757
  
  
  
  (gdb) b main
  
  Breakpoint 2 at 0x140013f7c: file C:/Users/bubaruch/Documents/projects/ffmpeg/ffmpeg/ffplay.c, line 3756.
  
  
  
  (gdb) r
  
  Starting program: C:\Users\bubaruch\Documents\projects\ffmpeg\b2\ffplay_g.exe
  
  [New Thread 15388.0x3278]
  
  ffplay version 2.8.git Copyright (c) 2003-2015 the FFmpeg developers
  
    built with gcc 5.2.0 (Rev4, Built by MSYS2 project)
  
    configuration: --disable-doc --disable-optimizations
  
    libavutil      55.  5.100 / 55.  5.100
  
    libavcodec     57. 15.100 / 57. 15.100
  
    libavformat    57. 14.100 / 57. 14.100
  
    libavdevice    57.  0.100 / 57.  0.100
  
    libavfilter     6. 15.100 /  6. 15.100
  
    libswscale      4.  0.100 /  4.  0.100
  
    libswresample   2.  0.101 /  2.  0.101
  
  Simple media player
  
  usage: ffplay [options] input_file
  
  
  
  An input file must be specified
  
  Use -h to get full help or, even better, run 'man ffplay'
  
  [Inferior 1 (process 15388) exited with code 01]