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• FFmpeg : Frame sizes of the generated video are extremely larger than the expected [closed]

  8 février 2021, par bbasaran

  I am recording frames (screen buffer) as NumPy arrays during the game which runs on the resolution of "400x225". Each frame array is a size of 270.1 kB.

  


  After saving those frames, I create an mp4 file with the following bash command (The game runs in 35 FPS (frames/second)) :

  


  ffmpeg -r 35 -f image2 -i frame%05d.png -vcodec libx264 -crf 1 video.mp4


  


  Then I have used a tool to generate a CSV file of frame data from the video created with the command above. The output is here below. The weird this is that, if we sum those first 35 frames (video was recorded with "-r 35" parameter because game runs in 35 FPS), we get approximately 18k kbit.

  


  18k kbit/sec bitrate is super high for a 400x225 video. What am I doing wrong while generating the video ? I appreciate any help, thanks !

  


  

  


• ffmpeg concatenated file does not play as expected

  28 janvier 2018, par Trewq

  I am trying create a video by concatenating in this order :

  video1.mp4 & video_from_static_image.mp4 & video1.mp4

  video1.mp4 is a simple video file. video_from_static_image.mp4 was created from the a.png image like this :

  ffmpeg  -y -framerate 25 -f image2  -loop 1 -i a.png -c:v libx264 -profile:v main -pix_fmt yuv420p -t 2 -r 25  video_from_static_image.mp4

  I create the final concatenated file through this command :

  ffmpeg -y -f concat -safe 0 -i mylist.txt -c copy final_concat.mp4

  I get the following message during the concat process :

  [mp4 @ 0x7fee6b815e00] Non-monotonous DTS in output stream 0:0; previous: 120678, current: 65722; changing to 120679. This may result in incorrect timestamps in the output file.

  The complete output when running the above command is here

  When I play final_concat.mp4 it has problems with video_from_static_image.mp4 and it seems to skip it (and other variations resulted in playing for a short while). I’ve tried a bunch of things, but I believe they all point to the DTS error above.

  BTW, all the video files are pretty small (< 1MB). Here are details about the version I am using :

  $ ffmpeg -version
ffmpeg version 3.3.4 Copyright (c) 2000-2017 the FFmpeg developers
built with Apple LLVM version 7.3.0 (clang-703.0.31)
configuration: --prefix=/usr/local/Cellar/ffmpeg/3.3.4 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libmp3lame --enable-libvpx --enable-libx264 --enable-libxvid --enable-opencl --enable-videotoolbox --disable-lzma --enable-vda
libavutil      55. 58.100 / 55. 58.100
libavcodec     57. 89.100 / 57. 89.100
libavformat    57. 71.100 / 57. 71.100
libavdevice    57.  6.100 / 57.  6.100
libavfilter     6. 82.100 /  6. 82.100
libavresample   3.  5.  0 /  3.  5.  0
libswscale      4.  6.100 /  4.  6.100
libswresample   2.  7.100 /  2.  7.100
libpostproc    54.  5.100 / 54.  5.100

• FFmpeg massive data loss when writing large data and swapping segments

  25 avril 2023, par Bohdan Petrenko

  I have an ffmpeg process running which continiously writes audio data to two 30 seconds (for testing, I'm actually planning to use 5 minutes) segments. The problem is that when I write some audio data with length more than size of two segments (60 seconds), 8-17 seconds of audio is lost. Here is how I run FFmpeg and write data :

  &#xA;

      _ffmpeg = Process.Start(new ProcessStartInfo
    {
        FileName = "ffmpeg",
        Arguments = 
            $"-y -f s16le -ar 48000 -ac {Channels} -i pipe:0 -c:a libmp3lame -f segment -segment_time {BufferDuration} -segment_format mp3 -segment_wrap 2 -reset_timestamps 1 -segment_list \"{_segmentListPath}\" \"{segmentName}\"",
        UseShellExecute = false,
        RedirectStandardInput = true
    })!;
    // Channels is usually 1, BufferDuration is 30


  &#xA;

  And here is how I write data :

  &#xA;

  public async Task WriteSilenceAsync(int amount)
{
    if (amount > _size) amount = _size; // _size is 48000 * 1 * 2 * 30 * 2 = 5760000 (size of 1 minute of audio)
    
    var silence = _silenceBuffer.AsMemory(0, amount);
    await _ffmpeg.StandardInput.BaseStream.WriteAsync(silence);
}


  &#xA;

  I tried to change the ffmpeg parameters and ways I write data. But I haven't found the solution.

  &#xA;

  I'm sure that the problem is caused by ffmpeg segments, because if I disable segmenting and write audio to a single file, there are no problems with data loss or audio missmatch. I also sure that amount of silence to add in WriteSilenceAsync() method is calculated right. I'm not sure if the problem appears with data length more than 30 seconds but less then 1 minute, but I think it doesn't.

  &#xA;

  I don't know how to solve this problem and would be glad to see any suggestions or solutions.

  &#xA;