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• Why does my webm not work in Unity but others do ?

  5 avril 2019, par UserDude

  Ok Im just trying to play a transparent video in Unity and Im following http://mikebrinkerink.com/2018/04/05/preparing-a-video-with-alpha-channel-for-unity/ and other tutorials.

  Ive tried exporting as ProRes 444 with alpha and as Webm VP8, and using terminal ffmpeg commands. Yet when I drop the video into Unity it loads indefinitely and I have to force quit.

  My file is this : https://drive.google.com/file/d/1RhOvYfeaK1rEaKdlTNcJB9EBpz4xxjvw/view?usp=sharing

  And this one that I downloaded works fine : https://simpl.info/videoalpha/video/dancer1.webm

  Whats the difference/how can I get mine to work ?

  Log :

  ffmpeg -i my_clip_with_alpha.avi -vcodec vp8 -acodec libvorbis -pix_fmt yuva420p -metadata:s:v:0 alpha_mode=”1″ -auto-alt-ref 0 my_vp8_clip_with_alpha.webm​

  as the command (with my clip’s name)

  Then

  built with Apple LLVM version 10.0.0 (clang-1000.11.45.5)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/4.1.2 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags='-I/Library/Java/JavaVirtualMachines/openjdk-11.0.2.jdk/Contents/Home/include -I/Library/Java/JavaVirtualMachines/openjdk-11.0.2.jdk/Contents/Home/include/darwin' --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libmp3lame --enable-libopus --enable-librubberband --enable-libsnappy --enable-libtesseract --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-videotoolbox --disable-libjack --disable-indev=jack --enable-libaom --enable-libsoxr
  
    libavutil      56. 22.100 / 56. 22.100
  
    libavcodec     58. 35.100 / 58. 35.100
  
    libavformat    58. 20.100 / 58. 20.100
  
    libavdevice    58.  5.100 / 58.  5.100
  
    libavfilter     7. 40.101 /  7. 40.101
  
    libavresample   4.  0.  0 /  4.  0.  0
  
    libswscale      5.  3.100 /  5.  3.100
  
    libswresample   3.  3.100 /  3.  3.100
  
    libpostproc    55.  3.100 / 55.  3.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Users/skylarthomas/Desktop/INSTRUCTIONS1_00000.mov':
  
    Metadata:
  
      major_brand     : qt  
  
      minor_version   : 537199360
  
      compatible_brands: qt  
  
      creation_time   : 2019-03-27T00:19:30.000000Z
  
    Duration: 00:00:17.05, start: 0.000000, bitrate: 40959 kb/s
  
      Stream #0:0(eng): Video: prores (ap4h / 0x68347061), yuva444p10le(tv, bt709, progressive), 2215x1773, 40467 kb/s, SAR 1:1 DAR 2215:1773, 29.97 fps, 29.97 tbr, 30k tbn, 30k tbc (default)
  
      Metadata:
  
        creation_time   : 2019-03-27T00:19:30.000000Z
  
        handler_name    : Apple Video Media Handler
  
        encoder         : Apple ProRes 4444
  
        timecode        : 00:00:00:00
  
      Stream #0:1(eng): Data: none (tmcd / 0x64636D74), 0 kb/s (default)
  
      Metadata:
  
        creation_time   : 2019-03-27T00:19:30.000000Z
  
        handler_name    : Time Code Media Handler
  
        timecode        : 00:00:00:00
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (prores (native) -> vp8 (libvpx))
  
  Press [q] to stop, [?] for help
  
  [libvpx @ 0x7fe11701d600] v1.8.0
  
  Output #0, webm, to 'my_vp8_clip_with_alpha.webm':
  
    Metadata:
  
      major_brand     : qt  
  
      minor_version   : 537199360
  
      compatible_brands: qt  
  
      encoder         : Lavf58.20.100
  
      Stream #0:0(eng): Video: vp8 (libvpx), yuva420p, 2215x1773 [SAR 1:1 DAR 2215:1773], q=-1--1, 200 kb/s, 29.97 fps, 1k tbn, 29.97 tbc (default)
  
      Metadata:
  
        creation_time   : 2019-03-27T00:19:30.000000Z
  
        handler_name    : Apple Video Media Handler
  
        timecode        : 00:00:00:00
  
        alpha_mode      : ”1″
  
        encoder         : Lavc58.35.100 libvpx
  
      Side data:
  
        cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
  
  frame=    2 fps=2.0 q=0.0 size=       1kB time=00:00:00.03 bitrate= 200.7kbits/sframe=    6 fps=3.7 q=0.0 size=       1kB time=00:00:00.16 bitrate=  40.6kbits/sframe=   10 fps=4.5 q=0.0 size=       1kB time=00:00:00.30 bitrate=  22.7kbits/s speedframe=   35 fps=2.9 q=0.0 size=       1kB time=00:00:01.13 bitrate=   6.0kbits/s speed=0.0934x    

• Is it possible to encode an audio stream with the exact length (to millisecond) ?

  27 avril 2019, par v010dya

  I am looking to create an output file with a length exactly 30 seconds in either mp3, aac, or vorbis codecs. I have used avconv (and also tried ffmpeg) with the following command :

  avconv  -i input_file_here -c:a aac -b:a 320k -t 30 out.mp4

  This produces the output, stating that it has encoded exactly 30 seconds, but then :

  $ avprobe out.mp4
  
  avprobe version v13_dev0-1648-gc4642788e, Copyright (c) 2007-2018 the Libav developers
  
    built on Apr 24 2019 15:49:35 with gcc 7 (Ubuntu 7.3.0-27ubuntu1~18.04)
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'out.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2mp41
  
      encoder         : Lavf57.83.100
  
    Duration: 00:00:30.02, start: 0.023220, bitrate: 323 kb/s
  
      Stream #0:0(und): Audio: aac (LC) [mp4a / 0x6134706D]
  
        44100 Hz, stereo, fltp, 321 kb/s (default)
  
      Metadata:
  
        handler_name    : SoundHandler
  
  # avprobe output

  When i encode in mp3 it correctly displays that it does more than 30 seconds during the encoding. When i try vorbis it does slightly less than 30 seconds.

  Is it possible to achieve the length of 30 seconds to the millisecond using any tool (i prefer either avconv).

• av_interleaved_write_frame() : Connection reset by peer - Using ffmpeg connected to local nginx

  10 avril 2019, par haxpor

  I have a local nginx web server installed with nginx-rtmp-module.

  I have 2 RTMP sources that are fed into local network at rtmp://127.0.0.1/live-video, and rtmp://127.0.0.1/live-audio. Then these twos will be combined and fed into rtmp://127.0.0.1/live again. The latter one will be the one that I will be using.

  The reason I did just that is I want to do noise reduction via sox.

  The setup is as follows

  1. rtmp://127.0.0.1/live-video - it uses ffmpeg to capture only screen (video) then output to such URL with command

     ffmpeg -analyzeduration 0 -video_size 1280x1080 -framerate 25 -f x11grab -i :0.0 -vcodec libx264 -flags +global_header -preset ultrafast -minrate 7200 -maxrate 8k -vsync 1 -f flv -metadata streamName=ZombieHeroLiveStream rtmp://127.0.0.1/live-video

  2. rtmp://127.0.0.1/live-audio - it uses sox to do noise reduction then uses ffmpeg to feed into such URL with command

      ffmpeg -f alsa -ac 1 -i default -preset ultrafast -f flac - | sox - -p noisered /tmp/noise.prof 0.21 | ffmpeg -analyzeduration 0 -thread_queue_size 2 -i - -f flv -preset ultrafast rtmp://127.0.0.1/live-audio

  3. rtmp://127.0.0.1/live - it uses ffmpeg to combine the above two sources then output into such URL, this is the one I tested viewing via VLC (which is works fine) with command

      ffmpeg -analyzeduration 0 -thread_queue_size 512 -rtmp_live live -flags +global_header -i "rtmp://127.0.0.1/live-video" -analyzeduration 0 -thread_queue_size 512 -rtmp_live live -flags +global_header -i "rtmp://127.0.0.1/live-audio" -c:v copy -c:a copy -map 0:v -map 1:a -async 1 -vsync 1 -video_size 1280x1080 -fflags +genpts -framerate 30 -preset ultrafast -minrate 7200 -maxrate 8k -flags +global_header -segment_list_flags +live -rtmp_live live -f flv -metadata streamName=ZombieHeroLiveStream "rtmp://127.0.0.1/live"

  The setup works fine, I tested and confirmed viewing via VLC targeting to URL of 3. The problem is that for some times after all threes above up and running, 1. will exit and thus make the whole encoding stop. The error shown is

  av_interleaved_write_frame(): Connection reset by peer29.88 bitrate=2021.8kbits/s dup=1791 drop=0 speed=0.998x    
  
      Last message repeated 2 times
  
  ...
  
  
  
  Error writing trailer of rtmp://127.0.0.1/live-video: Connection reset by peer

  No matter how I change and modify command line’s flags/options, it will end up like error above. Please note, I’ve modified flags quite a lot thus a certain flags might be possibly not needed.

  Full error log from 1. is here.

  Configuration of nginx server is here. It’s pretty much basic, but I will further use it to relay to multiple target RTMP servers later.

  I’m on Ubuntu 18.04, 4.18.0-17-generic with 8 GB of RAM, 4 CPU Cores with ffmpeg

  ffmpeg version 3.4.4-0ubuntu0.18.04.1 Copyright (c) 2000-2018 the FFmpeg developers
 built with gcc 7 (Ubuntu 7.3.0-16ubuntu3)

  , and with Nginx

  nginx version: nginx/1.14.2
  
  built by gcc 7.3.0 (Ubuntu 7.3.0-27ubuntu1~18.04) 
  
  built with OpenSSL 1.1.0g  2 Nov 2017 (running with OpenSSL 1.1.1b  26 Feb 2019)
  
  TLS SNI support enabled
  
  configure arguments: --with-http_ssl_module --add-module=../nginx-rtmp-module-1.2.1

  So again the question is how can I fix such issue ?
  Any suggestion would be appreciated.

  Important edit : I just found out I entered a wrong duplicated command for 3. Now it has been corrected ! Sorry about that.

  Update : I’m able to record audio with ffmpeg then do noise filter with sox, then stream to target rtmp with ffmpeg. But noise reduction doesn’t satisfy me yet. See above at 2. for its updated command. And according to this, sox has problem in understand a few bytes so I use flac format in piping instead. Still my question remained as this update is for improvement for command in 2.