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• FFMPEG use -strict -2

  21 août 2020, par GILO

  Hi I've been recently using the command below

  


  Ffmpeg -i song.mp3 -loop 1 -i image.jpg -filter_complex \
"[0:a]showfreqs=mode=line:ascale=log:fscale=log:s=1280x518[sf]; \
 [0:a]showwaves=s=1280x202:mode=p2p[sw]; \
 [sf][sw]vstack[fg]; \
 [1:v]scale=1280:-1,crop=iw:720[bg]; \
 [bg][fg]overlay=shortest=1:format=auto,format=yuv420p,drawtext=fontfile=/usr/share/fonts/TTF/Vera.ttf:fontcolor=white:x=10:y=10:text='\"Rated80s Prophets Prey\" by Comics On Film'[out]" \
-map "[out]" -map 0:a -c:v libx264 -preset fast -crf 18 -c:a libopus output.mp4 


  


  from this post Showfreqs and showwaves over background image ?.

  


  


  However I changed the output file to from output.mkv to output.mp4 as I would prefer to use mp4

  


  


  I get the below error

  


  [mp4 @ 0x7f962b816800] opus in MP4 support is experimental, add '-strict -2' if you want to use it.
Could not write header for output file #0 (incorrect codec parameters ?) : Experimental feature
Error initializing output stream 0:0 —
[libopus @ 0x7f962b819800] 1 frames left in the queue on closing

  


  The main information from that error is

  


  


  opus in MP4 support is experimental, add '-strict -2' if you want to use it.

  


  


  but where do I use -strict -2 within the command to make the command possible

  


• ffmpeg - add music to an audiobook, and loop the music [closed]

  23 novembre 2024, par Rhys

  I found these example, and they all work. But I cannot overlay 2 audios and loop the shortest audio until the longest audio is finished.

  


  Audio to match Video Length

  


  ffmpeg -i VIDEO1.mp4 -stream_loop -1 -i bgmusic.mp3 -shortest -c:v copy -c:a aac -map 0:v:0 -map 1:a:0 output.mp4


  


  Loop Audio

  


  ffmpeg -i audio.mp3 -filter_complex "aloop=loop=-1:size=10" output_loop.mp3


  


  Overlay 2 Audios

  


  ffmpeg -y -i bgmusic.mp3 -i audio.mp3 -filter_complex "[0:0]volume=1.0[a];[1:0]volume=1.0[b];[a][b]amix=inputs=2:duration=longest" -c:a libmp3lame output.mp3


  


  Here is my attempt.

  


  ffmpeg -y -i audio.mp3 -stream_loop -1 -i bgmusic.mp3 -shortest -filter_complex "[0:0]volume=1.0[a];[1:0]volume=1.0[b];[a][b]amix=inputs=2:duration=longest" -c:a libmp3lame output.mp3


  


  But this example is looping bgmusic.mp3 forever ... and audio.mp3 is stopping after the song finishes

  


  how can I get to audios to play togeather, but the shortest audio loops until the longest audio is finished ?

  


• FFmpeg - how to set output sample_size

  23 décembre 2019, par DennisJ

  Trying to create a simple command line player for .dsf (DSD audio) files, and output to an alsa device that supports up to 24-bit 192 kHz sample rate. The following command almost works and it does play the track. Examining the bold text below, the dsf input file is converted to 24-bit/192 kHz, but the output is then truncated to 16-bit 192 kHz (pcm_s16le i.e, 16 bit little endian).

  ffmpeg -i ’01 - Sweet Georgia Brown.dsf’ -f alsa hw:0,0

  After displaying the ffmpeg banner and song metadata (tags), here is the result, bold is my emphasis :

  Duration : 00:05:14.83, start : 0.000000, bitrate : 9234 kb/s
  Stream #0:0 : Audio : flac, 192000 Hz, stereo, s32 (24 bit)
  Stream mapping :
  Stream #0:0 -> #0:0 (flac (native) -> pcm_s16le (native))
  Press [q] to stop, [?] for help
  Output #0, alsa, to ’hw:0,0’ :

  Since I can play this and many other tracks at full resolution using another player (foobar2000) it seems there might be an option in the encoder which is part of FFmpeg : Lavf57.83.100 I can find no information in any of the FFmpeg documentation that helps. Tried finding options in FFplay and even guessing using other FFmpeg options like this example.
  ffmpeg -sample_fmt s24 -i ’01 - Sweet Georgia Brown.dsf’ -f alsa hw:0,0 ***** same results.

  I’m stuck. Any suggestions ?

  Environment : Linux Mint 19.2, 64-bit, ASUS Xonar STXii sound card.