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• Transcode any video to mp4 with max. 800 width or max. 800 height + watermark

  27 décembre 2018, par Mike

  I need a ffmpeg command that works with every video (with audio) format / type to encode it to h264 mp4. The output may have a maximum width of 800px and a maximum height of 800px. It would also be necesary to add a watermark in to bottom right corner... Is there a way to get all those things done with a single command line ? Even if it’s WMV, MOV, 3gp and whatever filetype is beeing used ?

• How to HLS-live-stream incoming batches of individual frames, "appending" to a m3u8 playlist in real time, with ffmpeg ?

  20 novembre 2024, par Rob

  My overall goal :

  



  Server-side :

  



  

  • I have batches of sequential, JPEG-encoded frames (8-16) arriving from time to time, generated at roughly 2 FPS.

  


  • I would like to host an HLS live stream, where, when a new batch of frames arrives, I encode those new frames as h264 .ts segments with ffmpeg, and have the new .ts segments automatically added to an HLS stream (e.g. .m3u8 file).

  


  



  Client/browser-side :

  



  

  • When the .m3u8 is updated, I would like the video stream being watched to simply "continue", advancing from the point where new .ts segments have been added.

  


  • I do not need the user to scrub backwards in time, the client just needs to support live observation of the stream.

  


  



  



  My current approach :

  



  Server-side :

  



  To generate the "first" few segments of the stream, I'm attempting the below (just command-line for now to get ffmpeg working right, but ultimately will be automated via a Python script) :

  



  For reference, I'm using ffmpeg version 3.4.6-0ubuntu0.18.04.1.

  



  ffmpeg -y -framerate 2 -i /frames/batch1/frame_%d.jpg \
       -c:v libx264 -crf 21 -preset veryfast -g 2 \
       -f hls -hls_time 4 -hls_list_size 4 -segment_wrap 4 -segment_list_flags +live video/stream.m3u8


  



  where the /frames/batch1/ folder contains a sequence of frames (e.g. frame_01.jpg, frame_02.jpg, etc...). This already doesn't appear to work correctly, because it keeps adding #EXT-X-ENDLIST to the end of the .m3u8 file, which as I understand is not correct for a live HLS stream - here's what that generates :

  



  #EXTM3U
#EXT-X-VERSION:3
#EXT-X-TARGETDURATION:4
#EXT-X-MEDIA-SEQUENCE:0
#EXTINF:4.000000,
stream0.ts
#EXTINF:4.000000,
stream1.ts
#EXTINF:2.000000,
stream2.ts
#EXT-X-ENDLIST


  



  I can't figure out how to suppress #EXT-X-ENDLIST here - this is problem #1.

  



  Then, to generate subsequent segments (e.g. when new frames become available), I'm trying this :

  



  ffmpeg -y -framerate 2 -start_number 20 -i /frames/batch2/frame_%d.jpg \
       -c:v libx264 -crf 21 -preset veryfast -g 2 \
       -f hls -hls_time 4 -hls_list_size 4 -segment_wrap 4 -segment_list_flags +live video/stream.m3u8


  



  Unfortunately, this does not work the way I want it to. It simply overwrites stream.m3u8, does and does not advance #EXT-X-MEDIA-SEQUENCE, it does not index the new .ts files correctly, and it also includes the undesirable #EXT-X-ENDLIST - this is the output of that command :

  



  #EXTM3U
#EXT-X-VERSION:3
#EXT-X-TARGETDURATION:4
#EXT-X-MEDIA-SEQUENCE:0
#EXTINF:4.000000,
stream0.ts
#EXTINF:4.000000,
stream1.ts
#EXTINF:3.000000,
stream2.ts
#EXT-X-ENDLIST


  



  Fundamentally, I can't figure out how to "append" to an existing .m3u8 in a way that makes sense for HLS live streaming. That's essentially problem #2.

  



  For hosting the stream, I'm using a simple Flask app - which appears to be working the way I intend - here's what I'm doing for reference :

  



  @app.route('/video/')
def stream(file_name):
    video_dir = './video'
    return send_from_directory(directory=video_dir, filename=file_name)


  



  Client-side :

  



  I'm trying HLS.js in Chrome - basically boils down to this :

  



  <video></video>&#xA;&#xA;...&#xA;&#xA;<code class="echappe-js">&lt;script src=&quot;https://cdn.jsdelivr.net/npm/hls.js@latest&quot;&gt;&lt;/script&gt;&#xA;&lt;script&gt;&amp;#xA;   var video = document.getElementById(&amp;#x27;video1&amp;#x27;);&amp;#xA;   if (Hls.isSupported()) {&amp;#xA;     var hls = new Hls();&amp;#xA;     hls.loadSource(&amp;#x27;/video/stream.m3u8&amp;#x27;);&amp;#xA;     hls.attachMedia(video);&amp;#xA;     hls.on(Hls.Events.MANIFEST_PARSED, function() {&amp;#xA;       video.play();&amp;#xA;     });&amp;#xA;   }&amp;#xA;   else if (video.canPlayType(&amp;#x27;application/vnd.apple.mpegurl&amp;#x27;)) {&amp;#xA;     video.src = &amp;#x27;/video/stream.m3u8&amp;#x27;;&amp;#xA;     video.addEventListener(&amp;#x27;loadedmetadata&amp;#x27;, function() {&amp;#xA;       video.play();&amp;#xA;     });&amp;#xA;   }&amp;#xA;&lt;/script&gt;   &#xA;

  



  I'd like to think that what I'm trying to do doesn't require a more complex approach than what I'm trying above, but since what I'm trying to far definitely isn't working, I'm starting to think I need to come at this from a different angle. Any ideas on what I'm missing ?

  



  Edit :

  



  I've also attempted the same (again in Chrome) with video.js, and am seeing similar behavior - in particular, when I manually update the backing stream.m3u8 (with no #EXT-X-ENDLIST tag), videojs never picks up the new changes to the live stream, and just buffers/hangs indefinitely.

  



  <video class="video-js vjs-default-skin" muted="muted" controls="controls">&#xA;    <source type="application/x-mpegURL" src="/video/stream.m3u8">&#xA;</source></video>&#xA;&#xA;...&#xA;&#xA;<code class="echappe-js">&lt;script&gt;&amp;#xA;    var player = videojs(&amp;#x27;video1&amp;#x27;);&amp;#xA;    player.play();&amp;#xA;&lt;/script&gt;&#xA;

  



  For example, if I start with this initial version of stream.m3u8 :

  



  #EXTM3U
#EXT-X-PLAYLIST-TYPE:EVENT
#EXT-X-VERSION:3
#EXT-X-TARGETDURATION:8
#EXT-X-MEDIA-SEQUENCE:0
#EXTINF:4.000000,
stream0.ts
#EXTINF:4.000000,
stream1.ts
#EXTINF:2.000000,
stream2.ts


  



  and then manually update it server-side to this :

  



  #EXTM3U
#EXT-X-PLAYLIST-TYPE:EVENT
#EXT-X-VERSION:3
#EXT-X-TARGETDURATION:8
#EXT-X-MEDIA-SEQUENCE:3
#EXTINF:4.000000,
stream3.ts
#EXTINF:4.000000,
stream4.ts
#EXTINF:3.000000,
stream5.ts


  



  the video.js control just buffers indefinitely after only playing the first 3 segments (stream*.ts 0-2), which isn't what I'd expect to happen (I'd expect it to continue playing stream*.ts 3-5 once stream.m3u8 is updated and video.js makes a request for the latest version of the playlist).

  


• When I append a silent audio (mp3) to an existing list of audio it garbles the final audio ?

  6 février 2020, par Marie

  After several hours I have narrowed down the issue with the garbled audio to be the 2-seconds silence audio mp3 I am appending (I think I had produced it once with Wavelab)

  However, I tried using ffmpeg according to a post to produce a similar 2 seconds audio but it too will corrupt/garble/chop voice in the final concatenation of audio files.

  ffmpeg -f lavfi -i anullsrc=r=44100:cl=mono -t 2 -q:a 9 -acodec libmp3lame SILENCE_2sec.MP3

  I typically will have several audio files to concatenate together but for simplicity I have able to narrow it to a couple of files simplifying to the following script. A simple Windows batch file you should be able to use and reproduce the issue at your end.

  rem 
  
  rem  
  
  SET EXE="S:\_BINS\FFmpeg 4.2.1 20200112\bin\ffmpeg.exe"
  
  
  
  SET ROOTPATH=.\
  
  
  
  SET IN_FILE="%ROOTPATH%MyList.txt"
  
  
  
  ECHO file '%ROOTPATH%HELLO.mp3' > MyList.txt
  
  ECHO file 'SILENCE_2sec.MP3' >> MyList.txt
  
  
  
  SET OPTIONS= -f concat -safe 0 -i  %IN_FILE%  -c copy -y
  
  
  
  SET OUT_FILE="%ROOTPATH%CONCATENATED_AUDIO_2.MP3"
  
  
  
  SET INFO_FILE="INFO.TXT"
  
  
  
  %EXE% %OPTIONS%  %OUT_FILE% 1> %INFO_FILE% 2>&1 
  
  
  
  ECHO ======================== >> %INFO_FILE%
  
  ECHO IN_FILE=%IN_FILE%  >> %INFO_FILE% 
  
  ECHO EXE=%EXE%  >> %INFO_FILE% 
  
  ECHO OPTIONS=%OPTIONS%  >> %INFO_FILE% 
  
  ECHO ======================== >> %INFO_FILE%

  Here is the console info output from the ffmpeg, let me know if you need other output include ones from ffprobe

  ffmpeg version git-2020-01-10-3d894db Copyright (c) 2000-2020 the FFmpeg developers
  
    built with gcc 9.2.1 (GCC) 20191125
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
  
    libavutil      56. 38.100 / 56. 38.100
  
    libavcodec     58. 65.103 / 58. 65.103
  
    libavformat    58. 35.101 / 58. 35.101
  
    libavdevice    58.  9.103 / 58.  9.103
  
    libavfilter     7. 70.101 /  7. 70.101
  
    libswscale      5.  6.100 /  5.  6.100
  
    libswresample   3.  6.100 /  3.  6.100
  
    libpostproc    55.  6.100 / 55.  6.100
  
  [mp3 @ 000000000036af80] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, concat, from '.\MyList.txt':
  
    Duration: N/A, start: 0.000000, bitrate: 32 kb/s
  
      Stream #0:0: Audio: mp3, 24000 Hz, mono, fltp, 32 kb/s
  
  Output #0, mp3, to '.\CONCATENATED_AUDIO_2.MP3':
  
    Metadata:
  
      TSSE            : Lavf58.35.101
  
      Stream #0:0: Audio: mp3, 24000 Hz, mono, fltp, 32 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  [mp3 @ 0000000000372d00] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 17280 >= 17255
  
  size=      11kB time=00:00:02.73 bitrate=  33.2kbits/s speed=2.73e+03x    
  
  video:0kB audio:11kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.137446%
  
  ======================== 
  
  IN_FILE=".\MyList.txt"   
  
  EXE="S:\_BINS\FFmpeg 4.2.1 20200112\bin\ffmpeg.exe"   
  
  OPTIONS= -f concat -safe 0 -i  ".\MyList.txt"  -c copy -y   
  
  ========================  

  I believe I am running FFmpeg 4.2.1, recently installed (20200112)

  You may produce the HELLO.mp3 by saving the following link

  https://translate.google.com.vn/translate_tts?en=UTF-8&q=Hello+&tl=en&client=tw-ob

  FYI, I am still a novice of ffmpeg and using it more like a black box with the help I received in this very super forum.
  Please be as explicit as you can with command line options on how I can fix this issue.
  Thank you.

  Additional Hints Debugging :

  If I append more files after the silence audio it seems that the silence audio impacts (garbles, chops) the previous audio.
  You may try the following for the list of audio files input.

  ECHO file '%ROOTPATH%HELLO.mp3' > MyList.txt
  
  ECHO file 'SILENCE_2sec.MP3' >> MyList.txt
  
  ECHO file '%ROOTPATH%HELLO.mp3' >> MyList.txt
  
  ECHO file '%ROOTPATH%HELLO.mp3' >> MyList.txt

  I typically add one or more silence file to derive a post silence effect after the actual audio. That’s my current logic. However if you have an alternative to appending a silence in the process of concatenating several audio files or appending x-seconds silence to an existing audio file. I can use that method as well from my coding.

  Thank you.