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• Converting mp4 to ogg file format results in a large file

  26 avril 2014, par parags

  I have a MP4 file of 83MB (converted from MOV of about 772MB using FFMPEG).
  For the file to be playable from all browsers from HTML5 video tag, I am converting the MP4 to OGG, again using FFMPEG command

  ffmpeg -i object-creation.mp4 -acodec libvorbis -vcodec libtheora -q:v 5 -q:a 5 object-creation-3.ogg

  The result of the above command is a very large OGG file of around 500 MB. I would certainly not want to upload such huge files to Amazon S3 (which I am using for storage, and distribution).

  Is there something I am missing here ? Is the file not compressed enough ?

  Is it possible to have the resultant file of somewhat manageable size like 80-100 MB without any appreciable loss in quality over what is seen in MP4 format ? Why is it that even the source file is 83MB, the resultant file is too big in comparison ?

  Thanks
  Parag

• Convert m4a to wav, but respect edit list ffmpeg [closed]

  4 février 2024, par user3413723

  I'm using ffmpeg to convert a m4a file to wav. The problem it the m4a file has an edts list that makes the audio start after a bit. I need this to be included so the wav is silent for that time. The audio has to be synced correctly. Any ideas ? I tried these :

  


  ffmpeg -i short.m4a  -acodec pcm_s16le -ac 1 -ar 16000 -use_editlist 1 out.wav

ffmpeg -i short.m4a  -acodec pcm_s16le -ac 1 -ar 16000 -advanced_editlist 0 out.wav


  


  no luck with either. Any ideas ? Here is the file

  


  https://drive.google.com/file/d/10x3QFjT_67ByfgXtiNFE1RlLAcwG68e5/view?usp=sharing

  


  Here is the debugger where you can see it adds time at the beginning. Just upload the m4a file and you can see the info about the audio track. You can also see it if you do ffprobe file.m4a like this :  Duration: 00:00:14.84, start: 0.167007, bitrate: 228 kb/s

  


  mp4 inspector :
https://gpac.github.io/mp4box.js/test/filereader.html

  


• I have a ffmpeg command to concatenate 300+ videos of different formats. What is the proper syntax for the concat complex filter ?

  25 avril 2022, par jokoon

  I plan to concatenate a large amount of video files of different formats and resolution, some without sound, and add a short black screen "pause" of about 0.5s between each.

  


  I wrote a python script to generate such command.

  


  I created a 0.5s video file using ffmpeg.exe -t 0.5 -f lavfi -i color=c=black:s=640x480 -c:v libx264 -tune stillimage -pix_fmt yuv420p blank500ms.mp4.

  


  I then added a silent audio to it with -f lavfi -i anullsrc -c:v copy -c:a aac -shortest

  


  I now have the problem of adding a blank audio track for streams without one, but I don't want to generate new file, I want to add it to my complex filter.

  


  This is my complex script and generate command.

  


  The command (there are line returns, because I send this with the python subprocess module)

  


  ffmpeg.exe
-i
input0.mp4
-i
input1.mp4
-i
input2.mp4
-i
input3.mp4
-i
input4.mp4
-i
input5.mp4
-i
input6.mp4
-i
input7.mp4
-i
input8.mp4
-i
input9.mp4
-i
input10.mp4
-f
lavfi
-i
anullsrc
-filter_complex_script
C:/filter_complex_script.txt
-map
"[final_video]"
-map
"[final_audio]"
output.mp4


  


  The complex_filter_script :

  


  [0]fps=24[fps0];
[fps0]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled0];
[1]fps=24[fps1];
[fps1]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled1];
[2]fps=24[fps2];
[fps2]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled2];
[3]fps=24[fps3];
[fps3]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled3];
[4]fps=24[fps4];
[fps4]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled4];
[5]fps=24[fps5];
[fps5]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled5];
[6]fps=24[fps6];
[fps6]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled6];
[7]fps=24[fps7];
[fps7]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled7];
[8]fps=24[fps8];
[fps8]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled8];
[9]fps=24[fps9];
[fps9]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled9];
[10]fps=24[fps10];
[fps10]scale=480:270:force_original_aspect_ratio=decrease,pad=480:270:(ow-iw)/2:(oh-ih)/2,setsar=1,setpts=PTS-STARTPTS[rescaled10];
[10]split=10[blank0][blank1][blank2][blank3][blank4][blank5][blank6][blank7][blank8][blank9];
[rescaled0:v][0:a][blank0][rescaled1:v][1:a][blank1][rescaled2:v][2:a][blank2][rescaled3:v][3:a][blank3][rescaled4:v][4:a][blank4][rescaled5:v][5:a][blank5][rescaled6:v][11:a][blank6][rescaled7:v][11:a][blank7][rescaled8:v][11:a][blank8][rescaled9:v][11:a][blank9]concat=n=22:v=1:a=1[final_video][final_audio]


  


  As you can see, some video use [11:a], because it's a silent audio stream.

  


  input10.mp4, mapped to [10] and then split (or "cloned") into blanked0 to 9, is a short pause separator.

  


  ffmpeg tells me the error

  


  [Parsed_split_55 @ 000001591c33b280] Media type mismatch between the 'Parsed_split_55' filter output pad 1 (video) and the 'Parsed_concat_56' filter input pad 5 (audio)
[AVFilterGraph @ 000001591bf1e6c0] Cannot create the link split:1 -> concat:5
Error initializing complex filters.
Invalid argument


  


  I'm a bit lost when it comes to using the [X:Y:Z] syntax, and how the order matter in the concat argument list.

  


  I'm open to any other suggestion to solve my problem. I would rather do this in a single command, without intermediate file.

  


  EDIT :

  


  For details, I already wrote a large concat+xstack filter that worked well with 8GB of memory.

  


  In this case, there are a lot of inputs, but those inputs are small, most of them are between 1 and 10MB, so it would probably not generate out-of-memory problems, although I'm not certain.