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• Hello I want to create a video from text using ffmpeg in my php script but i am not getting it

  28 décembre 2017, par Femzy

  I have been trying to solve this problem for a while now, but i have not got it through please someone should help me..
  My main objective is to collect user’n inputs and convert it to a video with black background, then the input text will be moving from left to right while the video is playing with audio in the background. Currently i was suggested to use "drawtext" but i have not find it working
  This is the code I use just to test the drawtext but the thing is, if i try it 10 times it will create a video file ones in many times i tried and even the video file would have 0 kb in size
  The Below code is my code

  echo shell_exec('ffmpeg -f lavfi -i testsrc=duration=5:size=800x600:rate=30 -vf drawtext="fontfile=/usr/share/fonts/truetype/dejavu/DejaVuSerif-Bold.ttf:textfile=text.txt: x=(w-tw)/2: y=h-(2*lh): fontcolor=white: box=1: boxcolor=0x00000000@1" -preset ultrafast outex.mp4');

• What's the Netflix / vmaf result stand for ?

  1er septembre 2021, par Hanamaki

  I ran the vamf between two mp4 files, and I got the result as :

  


  &#xA;<metric min="0.294617" max="0.999526" mean="0.902007"></metric>&#xA;<metric min="0.559920" max="0.994265" mean="0.918411"></metric>&#xA;<metric min="0.372149" max="0.999764" mean="0.907398"></metric>&#xA;<metric min="0.226668" max="0.999331" mean="0.915917"></metric>&#xA;<metric min="0.224996" max="1.010998" mean="0.881612"></metric>&#xA;<metric min="0.000000" max="32.177296" mean="6.365028"></metric>&#xA;<metric min="0.000000" max="145.765976" mean="7.202581"></metric>&#xA;<metric min="0.008498" max="0.910434" mean="0.247533"></metric>&#xA;<metric min="0.023310" max="0.990787" mean="0.457384"></metric>&#xA;<metric min="0.029596" max="0.995830" mean="0.529965"></metric>&#xA;<metric min="0.032303" max="0.997724" mean="0.586864"></metric>&#xA;<metric min="0.000000" max="100.000000" mean="71.630354"></metric>&#xA; &#xA;

  


  I got the "vif" from the paper "IMAGE INFORMATION AND VISUAL QUALITY". However,what's the meaning of 4 scales of vif ?
And any detail introduction about this result ? Thank you very much.

  


• Transcode HLS Segments individually using FFMPEG

  27 mai 2013, par rayh

  I am recording a continuous, live stream to a high-bitrate HLS stream. I then want to asynchronously transcode this to different formats/bitrates. I have this working, mostly, except audio artefacts are appearing between each segment (gaps and pops).

  Here is an example ffmpeg command line :

  ffmpeg -threads 1 -nostdin -loglevel verbose \
  
     -nostdin -y -i input.ts -c:a libfdk_aac \
  
     -ac 2 -b:a 64k -y -metadata -vn output.ts

  Inspecting an example sound file shows that there is a gap at the end of the audio :

  

  And the start of the file looks suspiciously attenuated (although this may not be an issue) :

  

  My suspicion is that these artefacts are happening because transcoding are occurring without the context of the stream as a whole.

  Any ideas on how to convince FFMPEG to produce audio that will fit back into a HLS stream ?

  ** UPDATE 1 **

  Here are the start/end of the original segment. As you can see, the start still appears the same, but the end is cleanly ended at 30s. I expect some degree of padding with lossy encoding, but I there is some way that HLS manages to do gapless playback (is this related to iTunes method with custom metadata ?)

  
  

  ** UPDATED 2 **

  So, I converted both the original (128k aac in MPEG2 TS) and the transcoded (64k aac in aac/adts container) to WAV and put the two side-by-side. This is the result :

  
  

  I'm not sure if this is representative of how a client will play it back, but it seems a bit odd that decoding the transcoded one introduces a gap at the start and makes the segment longer. Given they are both lossy encoding, I would have expected padding to be equally present in both (if at all).

  ** UPDATE 3 **

  According to http://en.wikipedia.org/wiki/Gapless_playback - Only a handful of encoders support gapless - for MP3, I've switched to lame in ffmpeg, and the problem, so far, appears to have gone.

  For AAC (see http://en.wikipedia.org/wiki/FAAC), I have tried libfaac (as opposed to libfdk_aac) and it also seems to produce gapless audio. However, the quality of the latter isn't that great and I'd rather use libfdk_aac is possible.