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• What is the most suitable H265 setting converting from H264 on ffmpeg ?

  9 juin 2017, par minion007

  Currently, I was trying to convert my videos no matter personal videos or movies from H264 to H265. But that was much harder than what I expected.

  Firstly, it took me over 10 hours to build ffmpeg on Linux and I believe I’m still having some problems on dealing with this.

  Secondly, I notice there is degradation after converting by using the setting or parameters from Internet or ffmpeg official website. On the other hand, the video size would be the same or even bigger than the original video size if I use small crf like 23 or 28.

  For instance, sudo ffmpeg -i input.mp4 -c:v hevc -x265-params lossless -preset medium -crf 30 -c:a copy output.mkv

  sudo ffmpeg -i input.mp4 -c:v libx265 -x265-params lossless -preset medium -crf 30 -c:a copy output.mkv

  the above two settings give me the very small video size (between 21% 23% of the original video) which is the main reason why I want to convert my videos to H265 but drawback would be the quality of converted video is worse than the original video (I would may be between 30% 35% degradation).

  Thirdly, the size of video would be much bigger than the original size of video if I use the lossless parameter.

  sudo ffmpeg -i input.mp4 -c:v libx265 -x265-params lossless=1 -preset medium -crf 30 -c:a copy output.mkv

  the above setting can keep the quality of the original video but size would be huge (above 200% of original video).

  Is there any suggestion ?

• FFMPEG replace video audio with filter_complex

  21 février 2017, par Chintan7027

  I wants such a output video where audio of output is created using ffmpeg -filter_complex mechanism,

  /usr/local/Cellar/ffmpeg/3.2.2/bin/ffmpeg 
  
  -i /uploads/videos/1487684390-lg9htt0RW2.mov 
  
  -i /uploads/audios/1487664761-SCPbo6Tkac.mp3 
  
  -filter_complex "
  
  [0:a]atrim=0:8.70824980736,asetpts=PTS-STARTPTS[aud1]; 
  
  [1:a]atrim=0:12.9567301273,asetpts=PTS-STARTPTS[aud2];
  
  [0:a]volume=0.3,atrim=start=8.70824980736:21.6649799347,asetpts=PTS-STARTPTS[slow_aud]; 
  
  [aud2][slow_aud] amerge=inputs=2[a_merged];
  
  [0:a]atrim=start=21.6649799347:31.6410098076 [remaining_audio]; 
  
  [aud1][a_merged][remaining_audio]concat=n=3:v=0:a=1[aout]" 
  
  -map 0:v -map "[aout]" -c:v copy -acodec mp3 
  
  /uploads/output/1487684390-lg9htt0RW2.mov

  Original Audio Recorded Based On UTC timestamp Vs Original Video Recorded Based on UTC timestamp

              13:00-------- Original Event Audio -------- 13:20
  
         12:50------------- Event Video Recorded --------------13:30

  This is my requirement

  So the audio of the output video should contains

  1. First 10 seconds(12:50 - 13:00) are Audio of Event Video Recorded
  2. Next 20 seconds (13:00 -13:20) are merged audio(Original Audio+ Original Video where Original video’s audio volume is .3)
  3. Remaining 10 seconds(13:21-13:30) of video will play remaing audio of video

  What I am getting by above commands

  1. First 10 seconds(12:50 - 13:00) are Audio of Event Video Recorded Achieved
  2. Next 20 seconds (13:00 -13:20) are merged audio(Original Audio+ Original Video where Original video’s audio volume is .3) Achieved
  3. Remaining 10 seconds(13:21-13:30) of video will play remaining audio of video Not Achieved

• How to convert audio files without changing bitrate using FFMPEG ? [duplicate]

  1er avril 2021, par Abraham Arnold

  I am converting MP3 file to WAV and I am using following code.

  


  ffmpeg -i input.mp3 -acodec pcm_s16le -ac 1 -ar 16000 output.wav


  


  So the original input.mp3 bitrate is 128kbps and when converted the output.wav file bitrate is 256kbps. I need the output file to be also keep same bitrate as original without changing it. I tried many ways. But I wasn't able to do that. I need some proper way to keep the original file bitrate value with output file. Anybody can help me ? Thanks in advance.