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• FFMPEG - Why doesn't this video converted by ffmpeg play on my LG TV while another does ?

  28 avril 2020, par RNP

  I have two files created from same big mp4 file. One (A.mp4) plays fine on my LG TV while the TV complains about the other (B.mp4). A.mp4 is of 5 seconds while B.mp4 is of 10 minutes.

  



  The first one was created with this ffmpeg command :

  



  ffmpeg -t 00:00:05  -i parent.mp4  -c:v h264 A.mp4


  



  When I used this command to convert the whole thing ( 2 and half hours), the resulting file didn't run on the tv. Only thing I could see different compared to A.mp4 was the large bit rate of 22000 kbps. Thinking I had found the problem, I ran the ffmpeg but restricted the bit rate to that of A.mp4. (Didn't use the codec setting since the parent is also h.264 encoded).

  



  ffmpeg -i parent.mp4 -t 00:10:00 -b:v 8540k B.mp4


  



  But this file didn't play on the TV. Both play fine on my Mac.

  



  Here is the diff report between the ffprobe output on two files :

  



  1c1
< Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'A.mp4':
---
> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'B.mp4':
7,8c7,8
<   Duration: 00:00:05.02, start: 0.000000, bitrate: 8653 kb/s
<     Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 8540 kb/s, 59.94 fps, 59.94 tbr, 60k tbn, 119.88 tbc (default)
---
>   Duration: 00:10:00.02, start: 0.000000, bitrate: 8642 kb/s
>     Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 8501 kb/s, 59.94 fps, 59.94 tbr, 60k tbn, 119.88 tbc (default)
11c11
<     Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 127 kb/s (default)
---
>     Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 128 kb/s (default)


  



  I would like to find right ffmpeg settings to convert the entire parent file so that it will play on the LG TV. Appreciate any ideas.

  


• ffmpeg with libopenh264 Is blurry [closed]

  11 septembre 2023, par Devin Dixon

  I am trying get live streaming using libopenh264 near the quality of libx264, but I am running into the quality issues. Here is a live stream standing still :

  


  

  


  And here is image when there is fast movement :

  


  

  


  The only documentation I've been able to find on options for libopenh264 is here : https://support.medialooks.com/hc/en-us/articles/360000210192-H-264-encoding-options

  


  Which give these values :

  


  





  


  


  


  


  


  



  


  


  


  


  


  


  


  


  


  


  


  


  


  


  


  


  


  


  


  


  


  


  


  


  


  Name	Default value	Description
  slices	0	Set the number of slices, used in the parallelized encoding. This is only used when slice_mode is set to ‘fixed’
  slice_mode	auto	Set slice mode.Possible values:fixedrowmbautodyn
  loopfilter	1	Enable loop filter, if set to 1. To disable set a value of 0.
  max_nal_size	0	Set maximum NAL size in bytes.
  allow_skip_frames	0	Allow skipping frames to hit the target bitrate if set to 1.

  


  


  But it seems options like slice_mode is incorrect, and playing around with options like slices and max_nal_size don't do anything. Here is my ffmpeg command :

  


  ffmpeg -y  -i pipe:0 -vf "scale=1920:1080:out_color_matrix=bt709" -color_primaries bt709 -color_trc bt709 -colorspace bt709 -c:v libopenh264 -b:v 9000k  -profile:v high -slices 10 -allow_skip_frames 1 -b_strategy 0 -maxrate 12000k -bufsize 9000k -pix_fmt yuv420p -r 60 -g 120 -c:a aac -b:a 256k -ac 2 -ar 44100  -af afftdn -f tee -map 0 "[f=flv]${data.stream.invirtu_rtmp_broadcast_endpoint}?sign=${data.stream.invirtu_rtmp_broadcast_key}|[f=mp4]${escapedTempFilePath}"`;


  


  Question :
  
Are there any other options I can be testing for libopenh264 to improve the quality ?

  


• Ffmpeg : Concat multiple videos and an audio file. Videos may or may not have audio and I cant know it before runnign the command

  28 juillet 2023, par AtActionPark

  I have a list of mp4 and an aac audio file. I have a concat.txt file with all the mp4s that need to be concatenated.
I need to concat all the mp4s and merge the aac on top.

  


  As said in the title, the mp4s individually may or may not have an audio track, and I can not check the mp4s with no audio, add a dummy sound track as it will be an automated process with thousands of commands executed daily.

  


  The sound of the mp4, if present, must be preserved and merged with the aac

  


  Here is a working command for the case where at least the first of my mp4s has an audiotrack :

  


  ffmpeg -f concat -safe 0 -i concat.txt -i audio.aac  -c:v copy -filter_complex "[0:a][1:a] amix=inputs=2:duration=shortest [audio_out]" -map 0:v -map "[audio_out]" -y result.mp4


  


  However that breaks if my mp4s dont have audio

  


  


  Stream specifier ':a' in filtergraph description [0:a][1:a]
amix=inputs=2:duration=shortest [audio_out] matches no streams.

  


  


  I've been reading about anullsrc, and would like to take [0:a] if it exists, or a generated empty audio channel, and merge it with my aac, but I'm not getting there.

  


  Any help would be appreciated

  


  Thanks