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• ffmpeg output file smaller than input file

  3 mai 2020, par Debug255

  I am using ffmpeg to rotate videos 90 or 180 degrees in a Python script. It works great. But, I am curious as to why the output file would be a smaller amount of bytes than the input file.

  



  Here are the commands I use :

  



  180 degrees :

  



  ffmpeg -i ./input.mp4 -preset veryslow -vf "transpose=2,transpose=2,format=yuv420p" -metadata:s:v rotate=0 -codec:v libx264 -codec:a copy ./output.mp4

  



  90 degrees :

  



  ffmpeg -i ./input.mp4 -vf "transpose=2" ./output.mp4

  



  For example, a GoPro Hero 3 MP4 file was originally 2.0 GB. The resulting output file was 480.9 MB. Another GoPro file was 2.0 and its resulting file was 671.5 MB. Is this maybe because the GoPro files were 2.0 but contains empty space, sort of like how some NTFS filesystems make a minimal 4k file, even when there is less bytes in it ?

  



  If this isn't the GoPro Hero 3, how do I rotate the files 90 or 180 degrees but ensure the output file size is the same ? Or, is data loss expected ? Does the data loss have to do with the format ?

  



  Note that the quality of the video doesn't appear to be damaged, which is good. So, I am interested in learning more about why this is happening, then I can read the section of ffmpeg documentation that is relevant to this.

  



  Thank you !

  


• How to stream the video from one PC to another with an acceptable quality and synchronization ?

  15 juin 2021, par ErickSkrauch

  I have the following task : to organize the broadcast of several gamers on the director's computer, which will switch the image to, to put it simply, the one who currently has more interesting gameplay.

  


  The obvious solution would be to raise an RTMP server and broadcast to it. We tried that. The image quality clearly correlates with the bitrate of the broadcast, but the streams aren't synchronized and there is no way to synchronize them. As far as I know, it's just not built into the RTMP protocol.

  


  We also tried streaming via UDP, SRT and RTSP protocols. We got minimal delay but a very blurry image and artifacts from lost packets. It feels like all these formats are trying to achieve constant FPS and sacrifice the quality.

  


  What we need :

  


  

  • A quality image.

  


  • Broken frames can be discarded (it's okay to have not constant FPS).

  


  • Latency isn't important.

  


  • The streams should be synchronized within a second or two.

  


  


  There is an assumption that broadcasting on UDP should be a solution, but some kind of intermediate buffer is needed to provide the necessary broadcasting conditions. But I don't know how to do that. I assume that we need an intermediate ffmpeg instance, which will read the incoming stream, buffer it and publish the result to some local port, from which the picture will be already taken by the director's OBS.

  


  Is there any solution to achieve our goals ?

  


• [ffmpeg C++ API] : How to copy a music file's cover image into another music file ?

  22 avril 2022, par Ananta

  I am having difficulties in copying a source music file's cover image into a destination music file. These two music files are in different formats (i.e, either mp3, flac, wav, or wma, different sampling rate). How should I implement the code for this ? I created a minimal code for this task below :

  


  const char* src_path = "source.mp3";
const char* dest_path = "destination.flac";
AVPacket src_pic; 

// open the source path
AVFormatContext *src_ctx = avformat_alloc_context();
avformat_open_input(&src_ctx, src_path, NULL, NULL);

// find the first attached picture, if available
for (i = 0; i < src_ctx->nb_streams; i++)
   if (src_ctx->streams[i]->disposition & AV_DISPOSITION_ATTACHED_PIC) {
      src_pic = src_ctx->streams[i]->attached_pic;

// open the destination path
AVFormatContext *dest_ctx = avformat_alloc_context();
avformat_open_input(&dest_ctx, dest_path, NULL, NULL);

// Then, how to embed the 'src_pic' into 'dest_ctx'?