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• using ffmpeg to create a wavefile image from opus

  29 décembre 2015, par edwardsmarkf

  I have been trying to use ffmpeg to create a wavefile image from an opus file. so far i have found three different methods but cannot seem to determine which one is the best.

  The end result is hopefully to have a sound-wave that is only approx. 55px in height. The image will become part of a css background-image.

  Adapted from Generating a waveform using ffmpeg :

  ffmpeg -i file.opus -filter_complex
  "showwavespic,colorbalance=bs=0.5:gm=0.3:bh=-0.5,drawbox=x=(iw-w)/2:y=(ih-h)/2:w=iw:h=1:color=black@0.5"
  file.png

  which produces this image :
  

  Next, I found this one (and my favorite because of the simplicity) :

  ffmpeg -i test.opus -lavfi showwavespic=split_channels=1:s=1024x800 test.png

  And here is what that one looks like :

  

  Finally, this one from FFmpeg Wiki : Waveform, but it seems less efficient using a second utility (gnuplot) rather than just ffmpeg :

  ffmpeg -i file.opus -ac 1 -filter:a
  aresample=4000 -map 0:a -c:a pcm_s16le -f data - | \
  gnuplot -e "set
  terminal png size 525,050 ;set output
  ’file.png’ ;unset key ;unset tics ;unset border ; set
  lmargin 0 ;set rmargin 0 ;set tmargin 0 ;set bmargin 0 ; plot ’

  

  Option two is my favorite, but i dont like the margins on the top and bottom of the waveforms.

  Option three (using gnuplot) makes the best ’shaped’ image for our needs, since the initial spike in sound seems to make the rest almost too small to use (lines tend to almost disappear) when the image is sized at only 50 pixels high.

  Any suggestions how might best approach this ? I really understand very little about any of the options I see, except of course for the size. Note too i have 10’s of thousands to process, so naturally i want to make a wise choice at the very beginning.

• ffmpeg messes up delayed inputs (repeats at the beginning)

  20 mars 2023, par Ralph

  ffmpeg plays audio inputs delayed to t1, t2 etc at t=0.
If there are several inputs, then it replicates one after the other (after in0 finishes with its duration duration(in0), plays another one at t=duration(in0), etc.)

  


  In the following example, the in0 is played at t=0.

  


  ffmpeg -i in0.mp3 -i in1.mp3 -i in2.mp3 -filter_complex "[0]adelay=1000[delayed1];[1]adelay=2000[delayed2];[2]adelay=3000[delayed3];[delayed1][delayed2][delayed3]amix=inputs=3:duration=longest" -codec:a libmp3lame -q:a 4 output.mp3


  


  The command performs the following : 1. The ffmpeg processes three input files (mp3). 2. The complex filter delays the inputs by 1, 2, 3 seconds input-wise. 3. amix mixes the delayed outputs together. 4. And finally, a coded transforms it into an output mp3.

  


  Another short version : 2 delayed beeps, an finally there come 3 beeps out (@ 0, 1, 2 seconds) :

  


  ffmpeg -i beep.mp3 -filter_complex "[0]adelay=1000[delayed1];[0]adelay=2000[delayed2];[delayed1][delayed2]amix=inputs=2:duration=longest" -codec:a libmp3lame -q:a 4 output.mp3


  


  ffmpeg version N-110011-gf456c192d9-tessus on a Mac M1, downloaded as binary from https://evermeet.cx/ffmpeg (01.2023).

  


  I added also a silent input file from 0 to 1 seconds without any alteration of the outcome.
I added a silent stream as input as well without improvement. E.g. :

  


  ... -filter_complex "aevalsrc=0:d=4[silence];...


  


  Another test with another ffmpeg version 5.1 Copyright (c) 2000-2022 the FFmpeg developers built with Apple clang version 13.1.6 (clang-1316.0.21.2.5) :
ffmpeg -i beep.mp3 -filter_complex "[0]adelay=1000[delayed1];aevalsrc=0:d=5[silence];[silence][delayed1]amix=inputs=2:duration=longest" -codec:a libmp3lame -q:a 4 output.mp3
There are again 2 beeps, at 0 and 1 second.

  


  (comment : concat solves the issue. But I want to have a flexible solution with possibly overlapping audio streams.) Thanks for giving a hint !

  


• ffmpeg convert rtmp audio/video stream to icecast2 audio/video stream

  17 avril 2023, par Solomon Thatcher

  I've been using this command to convert my public rtmp audio/video stream to a local mp3 audio icecast2 stream, but I have been unable to do the same for both video and audio.

  


  [Audio Only] (This works fine)
  
ffmpeg -re -i rtmp://162.142.xx.xxx:xxx/stream -vn -codec:a libmp3lame -b:a 128k -f mp3 -content_type audio/mpeg icecast://source:password@192.168.1.xxx:80/live

  


  I've tried to re-write in order to support video, but I keep hitting dead ends

  


  [Audio & Video Attempt] (this does not work)
  
ffmpeg -re -i rtmp://162.142.xx.xxx:xxx/stream -codec:v -f mpeg4 -b:v -f mpeg4 -content_type video/mpeg4 icecast://source:password@192.168.1.xxx:80/live

  


  When I run this command, it gives me the error below asking for a suitable format.

  


  $ ffmpeg -re -i rtmp://162.142.xx.xxx:xxx/stream -codec:v -f mpeg4 -b:v -f mpeg4 -content_type video/mpeg4 icecast://source:password@192.168.1.xxx:80/live
[h264 @ 0x5598ffbb8980] co located POCs unavailable
[h264 @ 0x5598ffbb8980] mmco: unref short failure
Input #0, flv, from 'rtmp://162.142.xx.xxx:xxx/stream':
  Metadata:
    |RtmpSampleAccess: true
    Server          : NGINX RTMP (github.com/sergey-dryabzhinsky/nginx-rtmp-module)
    displayWidth    : 1280
    displayHeight   : 720
    fps             : 48
    videokeyframe_frequency: 0
    profile         :
    level           :
  Duration: 00:00:00.00, start: 28117.779000, bitrate: N/A
  Stream #0:0: Audio: aac (LC), 48000 Hz, stereo, fltp, 327 kb/s
  Stream #0:1: Video: h264 (High), yuv420p(tv, bt709, progressive), 1280x720 [SAR 1:1 DAR 16:9], 2560 kb/s, 48 fps, 48 tbr, 1k tbn
[NULL @ 0x5598ffb8bec0] Unable to find a suitable output format for 'mpeg4'
mpeg4: Invalid argument


  


  I am positive that icecast2 can support video streams, however on the few occasions that I was able to actively stream successfully to it, it only showed an empty video embed.

  


  I've re-written the command for AV multiple times while referencing ffmpeg documentation, however my above attempt seems to be the closest (concept-wise) that I have gotten.

  


  What flags/formatting might I be missing which are causing the stream not to work ?