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• When creating ffmpeg segments instead of "-t" I cannot move inside the resulting MP4 file [closed]

  2 mai 2024, par pavlii

  We are using ffmpeg for grabbing X11 session and we had changed a bit the commant run. I will write only the different part of the command ; the rest were maintained so are not important for this problem :

  


  ffmpeg ..... -t 00:14:58 (old command)

  


  ffmpeg ..... -f segment -segment_time 900 -strftime 1 (new command)

  


  Resulting files are (in both cases) MP4. But in new - using segments - variant I am able to open the file in VLC player as usual but I cannot do any location inside. So I have to wait for example 700s to see what had happened at that time - I cannot use VLC slider to muve to that position.

  


  I am sure there is parametric solution I cannot find out yet :-)

  


  Many thanx for any ideas ; in case of further details no problem to add them.

  


  Regards, Pavel

  


  I am just browsing internet whether I find any tip for possible solution. I expected the behaviour like before, i.e. to be able to move to any location in resulting MP4 file when played back.

  


• avcodec/mips/aaccoder_mips : Remove MIPS-specific aaccoder

  15 mars 2024, par Andreas Rheinhardt

  avcodec/mips/aaccoder_mips : Remove MIPS-specific aaccoder
  ff_aac_coder_init_mips() modifies a static const structure of
  
  function pointers. This will crash if the binary uses relro
  
  and is a data race in any case.
  Furthermore it points to a maintainability issue : The
  
  AACCoefficientsEncoder structures have been constified
  
  in commit fd9212f2edfe9b107c3c08ba2df5fd2cba5ab9e3,
  
  a Libav commit merged in 318778de9ebec276cb9dfc65509231ca56590d13.
  
  Libav did not have the MIPS-specific AAC code and so this was
  
  fine for them ; yet FFmpeg had them, but this was not recognized.
  Commit 75a099fc734a4ee2b1347d0a3d8c53d883b95174 points to another
  
  maintainability issue : Contrary to ordinary DSP code, this code
  
  here is way more complex and needs to be constantly kept in sync
  
  with the ordinary code which it mimicks and replaces. Said commit
  
  is the only commit actually changing aaccoder.c in the last few
  
  years and the same change has not been performed for the MIPS
  
  clone ; before that, it even happened several times that the mips
  
  code was broken due to changes of the generic code (see commits
  
  97437bd17a8c5d4135b2f3b1b299bd7bb72ce02c and
  
  de262d018d7d7d9c967af1dfd1b861c4b9eb2a60 or
  
  860dbe0275e57cbf4228f3f653f872ff66ca596b or
  
  933309a6ca0f18bf1d40e917fff455221f57fb4b or
  
  b65ffa316e377213c29736929beba584d0d80d7c). This might even lead
  
  to scenarios where someone changing non-dsp aacenc code would
  
  have to modify mips inline asm in order to keep them in sync.
  
  This is obviously a significant burden (if the AAC encoder were
  
  actively developed).
  Finally, the code does not even compile here due to errors like
  
  "Error : float register should be even, was 1".
  Reviewed-by : Lynne <dev@lynne.ee>
  
  Reviewed-by : Jean-Baptiste Kempf <jb@videolan.org>
  
  Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
  

  • [DH] libavcodec/aacenc.c
  • [DH] libavcodec/aacenc.h
  • [DH] libavcodec/mips/Makefile
  • [DH] libavcodec/mips/aaccoder_mips.c

• Why every audio part louder in FFmpeg while I join them in one audio ?

  13 mai 2024, par Volodymyr Bilovus

  I trying to make dubbing for audio. I have original audio track and I want to put translated audio parts on top of the original.

  


  translated audio 100% vol : —p1--- ---p2— -----p3--- —p4—

  


  original audio 5% vol : -----------------------------------------

  


  Here is my FFmpeg command with filter_complex

  


  ffmpeg -i video_wpmXlZF4XiE.opus -i 989-audio.mp3 -i 989-audio.mp3 -i 989-audio.mp3 -i 989-audio.mp3 \
-filter_complex "\
[0:a]loudnorm=I=-14:TP=-2:LRA=7, volume=0.05[original]; \
[1:a]loudnorm=I=-14:TP=-2:LRA=7, adelay=5000|5000, volume=1.0[sent1]; \
[2:a]loudnorm=I=-14:TP=-2:LRA=7, adelay=10000|10000, volume=1.0[sent2]; \
[3:a]loudnorm=I=-14:TP=-2:LRA=7, adelay=20000|20000, volume=1.0[sent3]; \
[4:a]loudnorm=I=-14:TP=-2:LRA=7, adelay=30000|30000, volume=1.0[sent4]; \
[original][sent1][sent2][sent3][sent4]amix=inputs=5:duration=longest[out]" \
-map "[out]" output.mp3


  


  Audios I put on top of the original audio track is the same -i 989-audio.mp3 I made it by purpose to show the problem
And here is the audio levels on final generated track.


  


  As you can see, first and second only slightly different but third
and fourth have totally different(higher) volume level (Notice, audio is the same).
Why it's happened. And how can I workaround this odd behaviour ?