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• How do I accurately extend the duration of a tiny video clip ?

  18 avril 2018, par joeycato

  I noticed that ffmpeg doesn’t always scale the duration of a clip to the desired value ( even accounting for milliseconds of precision loss.) For example, whenever I attempt to scale the duration of this tiny 67ms clip to 7 seconds :

  ffmpeg -i input.avi -filter:v "setpts=(7/0.067)*PTS" -an output.avi    

  the resulting video is only 3.5 seconds in duration.

  Not really sure why this is off by multiple seconds, but I suspect it has something to do with the input clip itself ( which is only 2 frames @ 29.97 fps ) So far, I’ve discovered that if I scale the duration to a much larger value first, then scale back from that to 7 seconds, it works fine :

  ffmpeg -i input.avi -filter:v "setpts=(1000)*PTS" -an input_scaled_high.avi
  
  ffmpeg -i input_scaled_high.avi -filter:v "setpts=(7/33.400033)*PTS" -an output.avi

  But I’m hoping I don’t need to resort to an intermediate file. Is it possible to solve this with a single ffmpeg call ?

  If it helps, here is the video file I’m working with : https://drive.google.com/drive/folders/1hI5Xo6kfAfMd8ZylM6XrPX5fuO88gncE?usp=sharing

  Note : I should mention that I don’t need sound at all in this experiment ( hence the -an )

• How to stop a sound when certain other sound is inserted in the mix in ffmpeg ?

  3 avril 2022, par Antonio Oliveira

  I'm using a ffmpeg command that takes a set of sounds, mixes them into a single file, separating them by certain time intervals.

  


  Below is how my command is today.

  


  ffmpeg -i 
close_hh.wav    -i \
crash_l.wav     -i \
crash_r.wav     -i \
floor.wav       -i \
kick_l.wav      -i \
kick_r.wav      -i \
open_hh.wav     -i \
ride.wav        -i \
snare.wav       -i \
splash.wav      -i \
tom_1.wav       -i \
tom_2.wav       -i \
  tom_3.wav  -filter_complex  " [6]adelay=0|0[note_0];  [0]adelay=360|360[note_1];  [6]adelay=1260|1260[note_2];  [0]adelay=1537|1537[note_3];  [6]adelay=2494|2494[note_4];  [5]adelay=2767|2767[note_5];  [0]adelay=2969|2969[note_6];  [6]adelay=3673|3673[note_7];  [5]adelay=3924|3924[note_8];  [0]adelay=4132|4132[note_9];  [0][note_0][note_1][note_2][note_3][note_4][note_5][note_6][note_7][note_8][note_9]amix=inputs=11:normalize=0" record.wav


  


  This is the resulting audio that this command generates :

  


  ffmpg record.wav : https://drive.google.com/file/d/1LFV4ImLKLnRCqZRhZ7OqZy4Ecq5fwT3j/view?usp=sharing

  


  The purpose is to generate a drum recording, so I would like to simulate the dynamics of the hi-hat sounds : When the closed hi-hat is played, the open hi-hat will stop playing immediately if it is still sounding. The same behavior does not happen for any of the other sounds.

  


  One point that makes this a little more challenging is that other sounds can also be played between open hi-hat and closed hi-hat strikes, and theoretically the sound interruption behavior should work normally.

  


  Below is a recording demonstrating the expected result. (My app already reproduces the sound result I need internally, so I just made a simple recording with the microphone to illustrate)

  


  mic record.wav https://drive.google.com/file/d/19x19Fd_URQVo-MMCmGEHIC1SjaQbpWrh/view?usp=sharing

  


  Notice that in the first audio (ffmpeg record.wav) the first sound (open hi-hat) continues playing after the second is played.
In the second audio (mic record.wav) the first sound stops immediately after the second sound is played.

  


  How should the ffmpeg command be to get the expected result ?

  


• ffmpeg stream rc buffer underflow

  31 mai 2018, par finder2

  At the moment I’m setting up a screen-sharing platform with the opensource tool ffmpeg / ffserver. At the beginning of the sharing everything is fine. After around 1 1/2 minutes I get the following exception in the output.

  [flv @ 0x3a47aa0] rc buffer underflow
[flv @ 0x3a47aa0] max bitrate possibly too small or try trellis with large lmax or increase qmax

  I’ve tried to set an verry high lmax & qmax but this hasn’t changed any thing.Additionally I’ve tried to increase the bitrate , the buffersize and the bitrate.

  I use the following command with ffmpeg :

  ffmpeg -f x11grab -s 1920x1080 -r 20 -i :0.0+1680,0 "http://localserver.de:8080/input1.ffm"

  The config file for the ffserver is :

  HTTPPort 8080
  
  HTTPBindAddress 0.0.0.0
  
  MaxHTTPConnections 2000
  
  MaxClients 1000
  
  MaxBandwidth 50000
  
  CustomLog -
  
  
  
  <feed>
  
    File /var/ffserver/input1.ffm
  
    FileMaxSize 20M
  
    ACL allow *FROM IP*  *TO IP* 
  
  </feed>
  
  
  
  <stream>
  
    Feed input1.ffm
  
    Format swf
  
    VideoCodec flv
  
    VideoFrameRate 20
  
    VideoBufferSize 8000
  
    VideoBitRate 250
  
    VideoQMin 1
  
    VideoQMax 5
  
    VideoSize 640x400
  
    PreRoll 0
  
    StartSendOnKey
  
    NoAudio
  
  </stream>

  An other streaming format would also be a possibility but I didn’t know which are possible for live-streaming.