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• FFmpeg can't convert MOD to MP4

  7 août 2021, par Kaelan

  I'm trying to convert a MOD file to MP4, but it's not working. here is the command
  ffmpeg -y -i "Nintendo DSi + Internet.mod" "Nintendo DSi + Internet.mp4"

  And here is the log that it gives me
  

  


  ffmpeg version git-2020-02-13-6d37ca8 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9.2.1 (GCC) 20200122
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
  libavutil      56. 41.100 / 56. 41.100
  libavcodec     58. 68.102 / 58. 68.102
  libavformat    58. 38.100 / 58. 38.100
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 75.100 /  7. 75.100
  libswscale      5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
[mpeg @ 000002557deb95c0] Format mpeg detected only with low score of 25, misdetection possible!
[mp2 @ 000002557debb280] Header missing
    Last message repeated 18 times
[mpeg @ 000002557deb95c0] probed stream 1 failed
[mpeg @ 000002557deb95c0] probed stream 2 failed
[mp2 @ 000002557e395600] Header missing
    Last message repeated 59 times
[mp2 @ 000002557debb280] Header missing
[mp2 @ 000002557e395600] Header missing
[mpeg @ 000002557deb95c0] decoding for stream 0 failed
[mpeg @ 000002557deb95c0] decoding for stream 3 failed
[mpeg @ 000002557deb95c0] start time for stream 2 is not set in estimate_timings_from_pts
[mpeg @ 000002557deb95c0] stream 0 : no PTS found at end of file, duration not set
[mpeg @ 000002557deb95c0] stream 2 : no TS found at start of file, duration not set
[mpeg @ 000002557deb95c0] stream 3 : no PTS found at end of file, duration not set
[mpeg @ 000002557deb95c0] Could not find codec parameters for stream 0 (Audio: mp2, 0 channels, s16p): unspecified frame size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[mpeg @ 000002557deb95c0] Could not find codec parameters for stream 1 (Video: none, none): unknown codec
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[mpeg @ 000002557deb95c0] Could not find codec parameters for stream 2 (Video: none, none): unknown codec
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[mpeg @ 000002557deb95c0] Could not find codec parameters for stream 3 (Audio: mp2, 0 channels, s16p): unspecified frame size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Input #0, mpeg, from 'Nintendo DSi + Internet.mod':
  Duration: 02:14:02.94, start: 15898.332444, bitrate: 6 kb/s
    Stream #0:0[0x1c8]: Audio: mp2, 0 channels, s16p
    Stream #0:1[0x1ea]: Video: none, none, 90k tbr, 90k tbn
    Stream #0:2[0x1eb]: Video: none, none, 90k tbr, 90k tbn
    Stream #0:3[0x1d9]: Audio: mp2, 0 channels, s16p
Stream mapping:
  Stream #0:1 -> #0:0 (? (?) -> h264 (libx264))
  Stream #0:0 -> #0:1 (mp2 (native) -> aac (native))
Decoder (codec none) not found for input stream #0:1


  


  I got this mod file from the Nintendo DSi + Internet Video (by dumping it with godmode9i and getting the mod file using godmode9). Any help is appreciated.

  


• Streaming mp4a to localhost using udp and ffmpeg

  2 août 2017, par noswoscar

  I am using the following command to stream a video and it’s audio to localhost :
  ffmpeg -re -i out.mp4 -map 0:0 -vcodec libx264 -f h264 udp ://127.0.0.1:1234 -map 0:1 -acodec libfaac -f mp4a udp ://127.0.0.1:2020

  FFmpeg is not recognising my audio codec and my audio format so I get the following error message :
  Error

  What audio format and codec do I need to use ? The codec information of the video I wish to send is as follows :
  Codecs used

  When I convert the audio track to mp3 I can run the above command and stream the video and audio properly. However I dont want to convert all my video audio-tracks to mp3.

  (I am confused by all the encoders, decoders, codec names in the ffmpeg documentation) Is there a way of finding the right encoder to use with the mp4a audio codec other than reading the whole list of codecs and options ?

  Thanks.

• RTSP scene detection with ffmpeg

  3 septembre 2020, par Tobias

  I have a RTSP-stream and want to save individual images, whenever the scene is changing. The idea is actually to categorize these images manually and use them to train TensorFlow. I can easily grab the whole stream and generate thousands of images if I don't use the scene filter :

  


  ffmpeg -i rtsp://url -an -strftime 1 "%Y%m%d%H%M%S_scenechange.png"


  


  But as soon as I add the scene-filter ffmpeg is not saving any image, no matter how much the scene changes, the command is just extended by one parameter :

  


  ffmpeg -i rtsp://url -an -filter:v "select='gt(scene,0.1)'" -strftime 1 "%Y%m%d%H%M%S_scenechange.png"


  


  It starts ffmpeg with following output and then quits unexpectedly some minutes later :

  


  ffmpeg version git-2020-08-31-4a11a6f Copyright (c) 2000-2020 the FFmpeg developers
built with gcc 10.2.1 (GCC) 20200805
configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libgsm --enable-librav1e --enable-libsvtav1 --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
libavutil      56. 58.100 / 56. 58.100
libavcodec     58.101.101 / 58.101.101
libavformat    58. 51.101 / 58. 51.101
libavdevice    58. 11.101 / 58. 11.101
libavfilter     7. 87.100 /  7. 87.100
libswscale      5.  8.100 /  5.  8.100
libswresample   3.  8.100 /  3.  8.100
libpostproc    55.  8.100 / 55.  8.100
Input #0, rtsp, from 'rtsp://url':
Metadata:
title           : hysxrtpsion
Duration: N/A, start: 0.000000, bitrate: N/A
Stream #0:0: Video: h264 (High), yuv420p(progressive), 1920x1080, 15 fps, 15 tbr, 90k tbn, 30 tbc
Stream #0:1: Audio: aac (LC), 8000 Hz, mono, fltp
Stream mapping:
Stream #0:0 -> #0:0 (h264 (native) -> png (native))
Press [q] to stop, [?] for help
Output #0, image2, to 'mHS_scenechange.png':
Metadata:
title           : hysxrtpsion
encoder         : Lavf58.51.101
Stream #0:0: Video: png, rgb24, 1920x1080, q=2-31, 200 kb/s, 15 fps, 15 tbn,
15 tbc
Metadata:
  encoder         : Lavc58.101.101 png
[rtsp @ 00000000004c5b80] max delay reached. need to consume packet
[rtsp @ 00000000004c5b80] RTP: missed 1 packets
[rtsp @ 00000000004c5b80] max delay reached. need to consume packet
[rtsp @ 00000000004c5b80] RTP: missed 1 packets
frame=    0 fps=0.0 q=0.0 Lsize=N/A time=00:00:00.00 bitrate=N/A speed=   0x

video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)


  


  I looked at the mentioned options -ss -t -frames, but as you can see they're not used at all.

  


  Does anyone have more experience with that and could hint me in the right direction ? As the filter is the only single change I'm a bit lost here.